irclog2html for #asterisk on 20060524

00:00.15CrashHDanyone have any docs on regext? in sip.conf?
00:00.21CrashHDvoip-info won't let me access it
00:00.56*** join/#asterisk op3r (i=op3r@gr-153-202.eglobalreach.net)
00:03.47redondosQuestion: Everything works fine with my E200P card and the E1 line with PRI ISDN signalling. I can make/receive calls, everything is ok except that incoming calls don't have a caller ID. Should I talk to my provider about that or there exists the possibility that I need to configure something to enable it?
00:07.59*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)
00:09.24redondosanyone?
00:15.41*** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com)
00:17.40bigmac44442 PC's on a LAN.  1 can call, other gets 403 forbidden.  If shut down PC 1's phone the 2nd will then work?  What can i do about this?
00:18.15robl^change IP addresses.  sounds like maybe you have assigned the same IP to multiple devices
00:18.24pjchildsachandra, ser/openser's dispatcher appears to be stateless... so you aren't going to be able to track stuff like that...
00:19.04bigmac4444each PC has a unique LAN IP.
00:19.16bigmac4444both going out the same NAT
00:19.20pjchildsredondos, you can turn on pri debug span x and see if the incoming q931 has any CLI in it.. if not then talk to provider...
00:20.13redondospjchilds: Ohh, great information. Thanks.
00:20.53pjchilds~slap pjchilds
00:20.55jbotACTION slaps pjchilds, keep your grubby fingers to yourself!
00:21.35achandrapjchiilds: Yeah...I figured that based on what I read...as long as everything is up...its great... wondering if there is a uirky way of solving this using an external "check" of some sort..
00:22.21achandrapjchilds: sorry see my comment above..spelled your nick wrong
00:23.32*** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net)
00:23.41pjchildsachandra, no idea.. we use a stateful SER with three outbound * gateways, and just try each in turn based on the response (or lack of)..
00:24.13pjchildsachandra, we needed to do a bit of filtering, so if we get a 'error' that is a 'busy' or 'number not found' we just stop, rather than try on all three gateways...
00:25.11Flautowhat is the CUT function in asterisk?
00:25.13Flautowhat does it do
00:25.18Qwell[]Flauto: it cuts strings
00:25.30achandrapjchilds: when you say stateful SER how is that configured??
00:25.33Flautohow can it to be used? qwell
00:25.37Qwell[]show function CUT
00:25.47Flautothank your
00:25.57Flautowhatever i want to figure out, i can just use show?
00:26.02Dr-LinuxQwell[]: Digium support was also unable to help
00:26.20Qwell[]Dr-Linux: that sucks
00:29.02Dr-LinuxQwell[]
00:29.07Qwell[]Dr-Linux
00:29.09*** join/#asterisk watchy (n=watchy@h236.176.255.206.cable.cmdn.cablelynx.com)
00:29.17Dr-Linuxtoday first time i called the Digium support :)
00:29.32Dr-Linuxit was 1 hour long call , 40 minutes hold time and 20 talk time :)
00:29.35InfraReddid they tell you to FOAD?
00:29.44InfraRed:)
00:29.53tzangernah; that wouldn't take 20 minutes
00:29.54InfraRedhow much does it cost
00:29.55Dr-LinuxInfraRed: FOAD?
00:30.19InfraRedDr-Linux: http://www.urbandictionary.com/define.php?term=foad
00:30.30Dr-LinuxInfraRed: cost will be, if they understand my problem.
00:33.01pjchildsDr-Linux, what was the problem?
00:33.05pjchilds;)
00:33.40Dr-LinuxQwell[]: they said, these cards will work on 64bits/66Mhz slots, but it's not working
00:34.10Dr-Linuxhe accessed my server.
00:34.53Dr-Linuxpjchilds: i have 2 TE210P cards, but my system doesn't recognize them.
00:35.19tzangeryou should have traded them up to a single TE410
00:35.40Dr-Linuxhe used only command on my server "lspci -vv"
00:36.20Dr-Linuxtzanger: nope, that's according to our business requrements.
00:36.22achandraDr-Linux - did dmesg show anything?
00:36.31tzangerDr-Linux: did lspci -vv show two of htem?
00:36.39tzangeractually lspci  should show them
00:36.40Dr-Linuxachandra: anything what?
00:36.50achandrain terms of hardware loading
00:37.13pjchilds[root@adevrg05 asterisk]# lspci -vv | grep -i Xi
00:37.13pjchilds03:07.0 Communication controller: Xilinx Corporation Wildcard TE405P/TE410P (1st Gen) (rev 01)
00:37.16Dr-Linuxtzanger: nope, it it show, then there was no need to pissed off since 2 weeks
00:37.24achandrathe modules may be missing..but the hardware may show up on bootup
00:37.42achandrasimilar to wireless cards with no known modules...
00:37.43tzangerDr-Linux: yeah if the cards aren't enumerated by the pCI controller no driver in the world will find it
00:38.00Dr-Linuxachandra: if you even don't have asterisk installed, lspci -v should show the cards
00:38.33Dr-Linuxtzanger: but alteast they must have solution
00:38.34achandrayeah...im talking at the linux level...
00:38.48Dr-Linuxthis is not a solution to CHANGE THE SERVER
00:39.09achandrathats what they suggested??
00:39.14achandralol
00:39.19*** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net)
00:39.20tzangerDr-Linux: what is the solution?  sounds like a hardware problem (either the xilinx PCI block doesn't support 66mHz or there is something iffy about your controller
00:39.55achandramaybe kernel doesnt detect it??
00:40.06pjchildsis the server a HP 380 G4 ?
00:40.06Dr-Linuxachandra: nope
00:40.13tzangerachandra: again this is not something hte kernel is in control of
00:40.21Dr-Linuxtzanger: correct
00:40.31tzangereither the cards are enumerated when the PCI bridge is scanned or they aren't for one reason or another
00:40.31Dr-Linuxwell, guys
00:40.34achandrahmmm...interesting...
00:40.56tzangersounds like a corner case that Dr-Linux got snagged on
00:40.58Dr-Linuxhttp://206.111.151.217/dmesg
00:41.20Dr-Linuxif the httpd is running, you will be able to see all system specs, i put there
00:41.32pjchildsDr-Linux, 'connection refused...'
00:41.53Dr-Linuxwell, it's DELL PIII 550Mhz quad
00:42.27Dr-Linuxpjchilds: yeah, just datacenter support was changing the cards's slots
00:42.38Dr-Linuxso i boot the machine and didn't start the apache.
00:42.57Dr-Linuxtzanger: what you suggest?
00:43.11Dr-Linuxlooks like i have only 2 options.
00:43.16tzangerDr-Linux: do they show up in lspci output?
00:43.16Dr-Linux1. change the cards
00:43.23Dr-Linux2. change the machine. :(
00:43.35Dr-Linuxtzanger: nope, never show
00:43.51tzangeruse pastebin.ca and paste your lspci output (not -vv, just lspci)
00:44.04*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
00:45.10*** join/#asterisk MissNeBuN (n=whyban@user-12ld5f7.cable.mindspring.com)
00:45.34Dr-Linuxtzanger: oke wait
00:47.32Dr-Linuxhttp://pastebin.com/734295
00:48.47Dr-Linuxtzanger
00:49.09tzangerbroadcom PCI host
00:49.11tzangerthat's unusual
00:49.40tzangerI wonder ... Dr-Linux in the BIOS config is there any way to lock down PCI slot speeds or otherwise tinker with it?
00:50.07tzangerthat and all on one PCI bus too... that's not much of a server
00:50.23tzangermy dual xeon system has 7 PCI busses :-)
00:50.50tzangerhttp://pastebin.ca/58521
00:50.51Dr-Linuxtzafrir: even i called digium support and asked if i can decrease the PCI slot speeds, but he said, that's no possible :(
00:51.01tzangerDr-Linux: look in your BIOS
00:51.13tzangeryou never know what cockeyed things you can do until you poke around
00:51.21tzanger64bit is generally at least 64MHz though :-(
00:51.32Dr-Linux:S
00:52.15tzangerhowever I have a TE405 in a 3.3v slot on that motherboard I pastebinned and it works
00:52.20Snake-EyesIs there any way to stop asterisk from complaining about using inband dtmf when using g729 codec?
00:52.20Dr-Linuxtzanger: i also played with my BIOS, coudn't do anything without disabling some devices, like USB, floppy, etc
00:52.24tzangerso as I said, it could be something iffy with the BIOS config
00:52.34*** join/#asterisk dlynes_office (n=dlynes@216.251.149.66)
00:52.35tzangeryou can't adjust PCI timing or anything?
00:52.57Dr-Linuxtzanger: no, i can't
00:53.30Dr-Linuxtzanger: even i asked for to upgrade the BIOS to make that possible, but DELL support said, that's also not possible :(
00:53.50tzangerwell you may want to call Sangoma and ask them if their quadspan can do what you want... I know they are militant about getting their stuff to work in any system, but again -- I've never seen a broadcom PCI bridge before
00:53.54tzangerso I don't know
00:54.12tzangerIf Digium can't make it work then they can't make it work, what more can you ask
00:54.14Dr-Linuxtzanger: i think, i understand the exact problem, but just need to verify that from you.
00:55.00tzangerDr-Linux: basically when the kernel enumerates the PCI devices (when it does the PCI subsystem bringup) it's either not doing it right, or the cards are incompatible with the bus, so they don't get seen.
00:55.11tzangerDr-Linux: tell me, when you reboot the system (before linux boots) does the PCI summary list the devices?
00:55.58Dr-Linuxhhm..
00:56.13Dr-Linuxtzanger: not sure about that, or not understand your question :S
00:56.20tzangerif so, rebuild your kernel to use PCI BIOS accesses instead of direct and you may be able to get by
00:56.20Snake-Eyesinband works very nicly with are setup, where rfc2833 has problems
00:56.40*** join/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net)
00:56.43Dr-Linuxtzanger: if cards are plugged or not, but lspci shows devices
00:56.59tzangerDr-Linux: reboot the system (cold boot) -- BEFORE Linux boots the BIOS POST appears and MOST systems (Dell may have fucked you on this) present a "summary" screen just before it tries to boot. Do they show up in that summary screen?
00:57.22tzangerDr-Linux: I understand that.  Just reboot the system (cold boot) and tell me if you see them in the summary screen BEFORE linux boots (if you get a summary screen from your BIOS)
00:57.52Dr-Linuxawww
00:58.03Dr-Linuxit shows only few lines, not devices
00:58.11redondosWhat do you think of the Linksys PAP2-NA FXS adapter?
00:58.14redondosShould I get one?
00:58.21redondosThey are about 50 bucks.
00:58.23tzangerDr-Linux: ok
00:58.33syleworks fine for me
00:58.37dlynes_officeredondos: they're ok
00:58.42tzangerwell... recompile with PCI BIOS access only (not direct or any) and boot with the new kernel and see...
00:58.52Dr-Linuxtzanger: sorry friend i can't do that, bcoz i'm in pakistan and my server is located in the US datacenter, and they charge alot even for a small job :(
00:59.11tzangerif Digium has no other solutions (possible, but give them a chance), return the cards and talk to David Mandelstam at Sangoma.  Tell him Andrew from Listowel sent ya
00:59.27redondosThanks guys.
00:59.28sylewhy snake-eyes? you must be using cisco gateways somewhere then
00:59.29tzangerDr-Linux: well...  shame on you for not testing before shipping the system off...  not much else you can do then
01:00.44Dr-Linuxtzanger: maybe we will change the machine :S
01:00.55tzangerI mean think about it though, Dr-Linux, the system's *down* right now...  how much is it worth to try and get it up?
01:01.00Dr-Linuxtzanger: this machine have totall 7 slots
01:01.49Dr-Linuxtzanger: ok, but how can i re-biult the BIOS? if it's not possible as DELL support said
01:02.49Dr-Linuxtzanger: digium will not return my cards, but they will deduct 20 %
01:03.32tzangerI didn't say to rebuild the BIOS, I said to recompile the Linux kernel to use BIOS PCI accesses instead of Direct or Any PCI accesses.  Default is Any (i.e. try direct and fallback to BIOS) -- force BIOS only
01:03.52Dr-Linuxtzanger: and right now i came to home from work, my servers are not accesible from here, PIX is there
01:03.56tzangerDr-Linux: yes, I would too...  companies are not in the business of making your mistakes painless.  I'm not trying ot be a dick, just how it is
01:04.35Snake-Eyessyle, i think are pstn termination provider does
01:04.43tzangeraha
01:04.51Dr-Linuxtzanger: <tzanger> I didn't say to rebuild the BIOS, I said to recompile the Linux kernel to use BIOS PCI accesses instead of Direct or Any PCI accesses.  Default is Any (i.e. try direct and fallback to BIOS) -- force BIOS only
01:05.05tzangerDr-Linux: try booting the linux kernel with the pci=bios kernel parameter
01:05.17Dr-Linuxyour this point is valid, but i don't know how to do all this. :S but i'll try to read something about it first
01:05.41Dr-Linuxyes you are right
01:06.01tzangerDr-Linux: you can also try pci=conf1 or conf2
01:06.02dlynes_officeDr-Linux: you could always tell your boss to hire a local tech support person for California, too :)
01:06.06Snake-Eyessyle, ive used rfc and some menu systems i phone via pstn dont hear the dtmf, but when i use inband it all works fine, except asterisk compalins
01:06.07tzangerit may work, it may not work
01:06.13Dr-Linuxtzanger: i don't know how can i reboot linux kernel with the pci-bios kernel param...
01:06.38tzangerDr-Linux: well you are in a spot then, aren't you...  You need someone who can configure and administrate a linux box
01:06.44tzangerI think dlynes_office was offering
01:06.51dlynes_officetzanger: nope
01:06.53*** join/#asterisk op3r (i=op3r@gr-153-202.eglobalreach.net)
01:06.56dlynes_officetzanger: heh
01:07.16dlynes_officeI'm in British Columbia, not California :)
01:07.27tzangerdlynes_office: yeah but at least you're on the right coast.  I'm in Ontario :_)
01:07.33Dr-Linuxi dno't need one :)
01:07.35dlynes_officeheh
01:08.00Dr-Linuxtill yet we have sloved all kind of issue, hope will do so, we have 50 servers over there :S
01:08.12dlynes_officeDr-Linux: well, if it helps
01:08.38dlynes_officeDr-Linux: I know someone around here that speaks urdu, and can do system administration/sql adminning
01:09.35Dr-Linuxdlynes_office: dalnet is full of those type of guys :)
01:09.35dlynes_officeand he has no intention of moving back to Pakistan, so you don't have to worry about job security :)
01:09.41Dr-Linuxbut you guys are nice.
01:09.45Dr-Linuxthey are not
01:10.12b4kahey, anyone knows how to reset all the channels in a pri line? i have connected a pbx and it kinda wants all the channels reseted when i connect the cable
01:10.14Dr-LinuxURDU speakers never help other,
01:10.20b4kalike if i were the PSTN
01:10.25dlynes_officeDr-Linux: nah...this guy's a really nice guy
01:10.37dlynes_officeDr-Linux: i've known him since about 2 months after he arrived in Canada
01:11.04Dr-Linuxdlynes_office: i'm never worried :) have many options, but am dull according to your country and infront of you guys, but i'm good one according to my place :
01:11.11*** join/#asterisk cybergypsy (n=mark@APoitiers-152-1-92-194.w86-201.abo.wanadoo.fr)
01:11.15b4kaany command in asterisk to reset all the channels?
01:11.25dlynes_officeb4ka: restart when convenient
01:11.33Dr-Linuxif you say so..
01:12.01dlynes_officeb4ka: oh...just pri
01:12.09dlynes_officeb4ka: reload chan_zap.so
01:12.42b4kai have 2 pri lines
01:12.50b4kai dont want to reload all of them
01:12.55dlynes_officeb4ka: oh
01:12.57b4kathey t1 with the provider works
01:13.06b4kathe damnd pbx gets stuck sometimes
01:13.15b4kawe think the problem is it is waiting for the channels to reset
01:13.25b4kait suddenly stops sending data to the asterisk
01:13.38dlynes_officeb4ka: and you get busy signals dialing out on it and dialing in on it?
01:13.52dlynes_officeb4ka: and you might have 2 or 3 or 4 calls that never seem to get hung up?
01:14.10b4kakinda like that
01:14.22dlynes_officeb4ka: yeah...i experienced the same problem with zaptel 1.2.5
01:14.30b4kaoh?
01:14.36b4kaconnecting to a pbx?
01:14.45TripleFFFFFFFFFF[21:11] b4ka: any command in asterisk to reset all the channels? ?
01:14.45dlynes_officeb4ka: nah...just connecting to a regular pri
01:14.47TripleFFFFFFFFFFyeah
01:14.51TripleFFFFFFFFFFhit ctrl-c
01:14.53TripleFFFFFFFFFF;)
01:14.57TripleFFFFFFFFFFstop now
01:15.02TripleFFFFFFFFFFor shutdown;)
01:15.17TripleFFFFFFFFFFsoft hangup i assume you talking about ?
01:15.22b4kawell, i told you 1 pri works
01:15.25dlynes_officeb4ka: I grabbed libpri-trunk and zaptel-trunk, recompiled, reinstalled, and away i went
01:15.33dlynes_officeb4ka: i've never had a problem since
01:15.36Dr-Linuxb4ka: your connection is stable?
01:15.47b4kaits a crossover cable of 5mts...
01:15.49Dr-Linuxus soft hangup
01:16.14dlynes_officeDr-Linux: the whole port dies on him
01:16.14b4kasometimes i can make calls
01:16.20dlynes_officeDr-Linux: erm the whole span
01:16.21b4kathen it suddenly dies
01:16.26b4kayeah
01:16.28b4kaand only THAT span
01:16.38Dr-Linuxb4ka: if the call is brigded and connection breaks, the the channels hangs like that
01:16.39dlynes_officeb4ka: usually after about 22 hours or more up time, right?
01:16.48b4kano
01:16.53b4kalike 10 minutos
01:16.55b4kaminutes
01:16.56Dr-Linuxi see
01:16.57dlynes_officedamn
01:17.04b4kaand when i restart sometimes it doesnt work
01:17.13dlynes_officeyeah...for me it was like after three weeks the first time
01:17.23dlynes_officeabout 1-1/2 to 2 weeks the second time
01:17.32dlynes_officeabout 4 or 5 days the next time
01:17.40dlynes_officethen about every 2 or 3 days for a little while
01:17.40Dr-Linuxhhm..
01:17.50Dr-Linuxbut i think that could be connection problem.
01:17.51dlynes_officeand then every day or two
01:18.12b4kaill pastebin some crap
01:19.17Dr-Linuxb4ka: i was facing some problem. but mine was connection problem, i found later
01:19.31b4kathat happens when i make a call now
01:19.35b4kahttp://pastebin.com/734339
01:19.50b4kaDr-Linux: its a 5mt crossover t1 cable
01:20.07b4kaand sometimes it works, so its not a bad cable
01:20.18dlynes_officeyeah...looks like the same problem i was having
01:20.26dlynes_officebut that's just from your log
01:20.31Dr-Linuxb4ka: it's directly connection, or you have local network?
01:20.36dlynes_officeyour log could be caused by just about anything
01:20.52b4kadirect connection
01:20.59b4kathey are side by side
01:21.11Dr-Linuxb4ka: what asterisk version?
01:21.24b4ka1.2.5
01:21.39b4kaerrrr
01:21.42b4ka1.2.7.1
01:22.15*** part/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net)
01:24.16b4kaasterisk-usa*CLI> pri show span 2
01:24.17b4kaPrimary D-channel: 48
01:24.17b4kaStatus: Provisioned, Down, Active
01:24.30b4kai cant figure out why its down
01:24.47dlynes_officezap show status?
01:24.55dlynes_officedo you see a yellow alarm?
01:25.35Flautohey, dlynes
01:25.37Flautohow are you doing
01:25.40dlynes_officegood
01:25.50Flautoi was reading for call record
01:25.54dlynes_officewas just getting ready to head out
01:26.00Flautoanything good that i can read?
01:26.01dlynes_officereading for call record?
01:26.05Flautoyes
01:26.08dlynes_officeyou mean cdr?
01:26.14Flautono
01:26.16dlynes_officeor recording a conversation?
01:26.19b4kano alarms
01:26.24Flautoyes
01:26.28dlynes_officeapp MixMonitor()
01:26.40Flautohow would i use it
01:26.48dlynes_officemake sure you're using asterisk 1.2.7.0 or higher, too
01:26.53Flautoyes
01:26.57Flautoi have 1.2.7.1
01:26.58dlynes_officeotherwise you might find it'll segfault on you
01:27.29Flautolet me search for mixmonitor online then
01:27.58dlynes_officeThere's full documentation on it including examples under voip-info.org->asterisk pbx->applications->recording/playback->MixMonitor()
01:28.53Dr-Linuxi'm using Monitor() app with sox package to mix
01:29.22dlynes_officeDr-Linux: mixmonitor doesn't need to spawn a separate process to mix the call legs
01:29.35b4kadlynes_office
01:29.38*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)
01:29.48b4kawhat did you do to solve your problem? i dont think its the same but wth..
01:29.54b4kaat this point ill try anything
01:29.55Dr-Linuxdlynes_office: yes, but i'm using old version 1.2.0
01:30.34*** join/#asterisk zotz (n=zotz@24.231.36.9)
01:30.37Flautookay, thanks. dlynes
01:31.19*** part/#asterisk downunder33 (n=robert@219.95.251.17)
01:31.44CrashHDhey dlynes_office, can I pick your brain a bit?
01:32.06dlynes_officeb4ka: i installed libpri-trunk and zaptel-trunk
01:32.24dlynes_officeb4ka: they're always in a state of flux though...i just got lucky to get versions that worked properly
01:32.55dlynes_officeb4ka: if you want, i can put my svn snapshots up for http for you
01:32.59dlynes_officeCrashHD: what's up?
01:33.09dlynes_officeCrashHD: smack
01:33.54CrashHDdlynes_office: newbie I need to find a solution for displaying the extension on the phone, but having more than one of the same extension on the system (sip.conf ext regs)
01:33.55dlynes_officeb4ka: i've been running that particular trunk version for about 2-1/2 months now without any more issues
01:34.10*** join/#asterisk Sponge_bob (n=None@cpe-66-27-162-13.socal.res.rr.com)
01:34.19CrashHDdlynes_office: same stuff we were dealing with yesterday
01:34.55dlynes_officeCrashHD: You mean both the extension name and the extension number?
01:35.16CrashHDin my sip.conf I basically need to have two [221]'s
01:35.25CrashHDso that I can use those settings on different phones
01:35.32CrashHDand that text will be displayed on the phone
01:35.42dlynes_officeCrashHD: answer the question :)
01:36.04CrashHDplease rephrase
01:36.12b4kawhat is zaptel-trunk?
01:36.22dlynes_officeCrashHD: i.e. show both 221 and Reception on the phone at the same time, when it's idle, right?
01:36.36dlynes_officeb4ka: it's the unstable branch of the the zaptel drivers
01:36.44dlynes_officeb4ka: i.e. the development trunk
01:36.57dlynes_officeb4ka: it's what will eventually become 1.2.8 and/or 1.4
01:37.06CrashHDdlynes_office: currently all phones show the User ID that was used to register the line
01:37.17dlynes_officeb4ka: erm i mean 1.2.6 and and/or 1.4
01:37.33CrashHDdlynes_office: I need to be able to use the same user ID twice (221 x 2)
01:37.42dlynes_officeCrashHD: but your user id and your caller id are the same value, right?
01:37.45IceManRISKanyone here already use jiax ?
01:38.11CrashHDon the astra ya
01:38.25dlynes_officeCrashHD: try changing the caller id; see if that changes the display
01:38.31CrashHDdoes
01:38.48dlynes_officeCrashHD: then what's the problem?
01:38.53CrashHDthe display is set from the Phone number field
01:38.57Sponge_bobi have a tdm400p card. does anyone know why when i call into the fxo port from 'outside' it sometimes picks up and sometimes just instantly drops the calls?
01:39.00CrashHD*it doesn't I mean
01:39.01CrashHDsorry
01:39.07CrashHDhit enter mid-sentence
01:39.30CrashHDthe phone number field relates to the [sip_heading]
01:39.42dlynes_officeCrashHD: ok...go on
01:39.45CrashHDwhich seems to have to match the username=var
01:39.55CrashHDso to have a context of [221]
01:39.56dlynes_officeCrashHD: are you sure?
01:40.31CrashHDdlynes_office: I registered a phone fine, changed the username=to_something_else
01:40.35CrashHDand it gave me an error
01:40.46CrashHDand tried all possibilties of combo's to use
01:40.50CrashHDto get it to reg
01:40.59CrashHDsetting the auth Name to the new username value
01:41.09CrashHDand vise versa with the phone number, auth name field
01:41.21dlynes_officeyeah...i don't know what to tell you then
01:41.26CrashHDheh ok
01:41.30CrashHDI picked away
01:41.36CrashHDknow anyone who may be able to assist?
01:41.42dlynes_officei'm not going to have a chance to work on the aastras again for about 5 or 6 days
01:41.47CrashHD16 hours on this one stupid thing seems retarded
01:41.51dlynes_officetry Ciber311 if you see him around
01:41.59dlynes_office~seen ciber311
01:42.06jbotciber311 <n=Ciber@user-1087e94.cable.mindspring.com> was last seen on IRC in channel #asterisk, 16d 23h 41m 13s ago, saying: 'well afk a bit'.
01:42.09zwelchCrashHD: one thought is that the phone isn't let you set your phone number/authname separately
01:42.13dlynes_officeok, guess not
01:42.32*** join/#asterisk Gabriel25 (n=whyban@user-12ld5f7.cable.mindspring.com)
01:42.33CrashHDzwelch: 3 seperate phone models result in the same behavior
01:42.34zwelchi.e. that's why you have to have them be the same
01:42.56dlynes_officeCrashHD: write your own firmware for them, then!
01:43.00CrashHDdlynes_office: hehe, well if you do think of anyone I would appreciate the assistance
01:43.03zwelchheh, well, i reached the same conclusion that you did; the [xx] has to equal username=xx
01:43.18zwelchbut that's a problem for your scenario, i would guess
01:43.20dlynes_officeCrashHD: or run asterisk on virtual hostnames :)
01:43.27dlynes_officeor virtual ips
01:43.45CrashHDzwelch: but what is the additional setting on the phones for, there are three auth fields one is password and two are usernames of sorts
01:43.47dlynes_officeanyways
01:43.50dlynes_officei've gotta run
01:43.57dlynes_officecatch y'all later
01:43.58CrashHDthank dlynes_office
01:43.59CrashHDttyl
01:44.09Supaplexhehe
01:44.28zwelchCrashHD: i'm not sure; i'm just speculating
01:44.32CrashHD*nods*
01:44.39zwelchbut i really do have to run off to, just know that you're not alone ;)
01:44.56CrashHDso can someone explain what the three different auth fields in sip are for and if they are implemented in asterisk?
01:54.39*** join/#asterisk mogorman (n=mogorman@68.62.237.103)
01:55.15bigmac4444still have problems using extra VoIP phones behind the same public IP, using NAT :S
01:58.57*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
02:01.41*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
02:02.13*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
02:06.46CrashHDhow can we make the extension numbers not be displayed on the phones when the register?
02:06.53CrashHDall phones do this and it is driving me nuts
02:07.32CrashHDwell the 4 I have tried today and yesterday do anyway
02:08.01Gabriel25X-Asterisk-HangupCause: No route to destination
02:08.12Gabriel25how I can fix this ?
02:08.14CrashHDsip or iax?
02:08.19Gabriel25SIP
02:08.26CrashHDyou have the noreinvite off?
02:08.45Gabriel25I have no idea whats that
02:08.57Gabriel25where I can check that
02:08.58Gabriel25?
02:09.39CrashHDin your sip.conf [general]
02:09.48CrashHDtry doing canreinvite=no
02:09.58Gabriel25ok
02:10.19CrashHDthis will keep asterisk in the media stream (if that is the problem the error message is talking about)
02:10.21CrashHDI could be wrong
02:11.27Gabriel25here is my sip.conf
02:11.29Gabriel25http://pastebin.ca/58535
02:11.52CrashHDwhat's in sip_nat.conf?
02:11.56CrashHDand the other files?
02:12.06*** join/#asterisk rabelais (n=blank@santamonica-cuda4-24-55-43-249.vnnyca.adelphia.net)
02:12.28[TK]D-FenderYay more AMP!
02:12.37Gabriel25sip_additional.conf  sip.conf             sip_notify.conf
02:12.46CrashHDAMP
02:14.30Gabriel25which one CrashHD ?
02:15.00CrashHDsip_nat
02:15.20CrashHDwhat kind of call is this on?
02:15.25CrashHDlan to lan?
02:15.27CrashHDlan to wan?
02:15.31CrashHDwan to wan?
02:15.35Flautohow do i use mixmonitor
02:15.43QwellFlauto: show application mixmonitor
02:15.44CrashHD~mixmonitor
02:17.45Gabriel25LAN
02:18.49Flautoqwell, do i need to set an extension in my dialplan or something else
02:21.38[TK]D-Fenderfile : You're rminding me of my favourite russian author : Imaknockoff ;)
02:21.49filehahahahaha
02:22.33*** join/#asterisk fugitivo (n=ajf@201.216.246.181)
02:22.35fugitivohi
02:22.59fugitivoi'm having serious issues with my digium cards
02:23.09fugitivomy zttest is crap
02:23.16CrashHDcall digium, they love to support their equipment
02:23.16fugitivono interrupt sharing at all
02:23.21[TK]D-Fenderfugitivo : Sharing IRQ's?
02:23.29fugitivo[TK]D-Fender: no
02:23.30[TK]D-Fenderfug, pastebin a dmesg.
02:23.31fileincompatibility motherboard?
02:23.47fugitivofile: it's not in the list of incompatibilities
02:24.00fugitivowhen i don't use a network card at all
02:24.02fugitivoit works ok
02:24.17fugitivoif I use a network card, zttest starts to fail
02:24.19filewhat network card?
02:24.28fugitivoe1000
02:24.30[TK]D-Fendere100 !
02:24.35Qwelle10000!
02:24.42[TK]D-Fenderfugitivo : Thats a super-no-no for Digium cards...
02:24.52fugitivoreally?
02:25.01Qwellyeah..really?
02:25.12[TK]D-Fenderfugitivo : And the best part, its standard issue on a the MAJORITY of serverboards out there...
02:25.23fugitivowhat a crap
02:25.23fileactually, just for kicks...
02:25.24CrashHDwhy is it a no no?
02:25.25CrashHDtechnically?
02:25.27[TK]D-Fenderfugitivo : Yes, I had one in mine and let me tell you.... all helll...
02:25.41filesee if you can compile the E1000 driver into the compile instead of a module...
02:25.43fugitivook, so if i change the network card, it could work?
02:25.43[TK]D-Fenderinterrupt load makes Digium cards jealous ;)
02:25.50fugitivofile: i did it
02:25.56fileah, then yeah...
02:26.02fugitivofile: it works better, but fails sometimes
02:26.11fileI'm a Broadcom person myself
02:26.13[TK]D-Fenderfugitivo : Make sure the driver module doesn't even LOAD, and get yoursef a PCI NIC.
02:26.17Qwellrealtek!
02:26.34fugitivo[TK]D-Fender: i have a pci nic and onboard nic, both e1000 :)
02:26.56[TK]D-Fenderfugitivo : Good work!
02:27.08CrashHDe1000 or e100 is the problem (or both)?
02:27.22[TK]D-Fenderfugitivo : And next you're going to tell me you're on an i7505 chipset MB too.....
02:27.26[TK]D-Fendere1000
02:27.31fugitivoi'll stole a nic from the firewall
02:27.33[TK]D-FenderI typo'd
02:28.01CrashHDare the sanogma cards as picky about hardware?
02:28.17[TK]D-Fender'course thanks to some nudging from file here I get to run whatever the hell I want in my server now without worry ;)
02:28.28[TK]D-FenderCrashHD : Not at all.
02:28.44fugitivo[TK]D-Fender: i think it's not, the mb wasn't on the incompatibility list
02:28.49filethe Sangoma ones aren't as picky, but there's still some hardware out there that they have the same issue...
02:29.05*** join/#asterisk iq|mobile (n=iq@71-215-34-237.omah.qwest.net)
02:29.38[TK]D-Fenderfugitivo : Yeah well your NIC is.
02:29.55[TK]D-Fenderfugitivo : the e1000 has been a star offender for years now.
02:30.10[TK]D-Fenderfile : Really?  What have you run into?
02:30.28fugitivogreat, the firewall is using 2 3com
02:30.32fugitivo3com is ok?
02:30.39file[TK]D-Fender: I remember reading on the mailing list about someone having issues with specific hardware... had to go to Sangoma to get it resolved via firmware upgrade I believe
02:30.41fugitivo3c905C?
02:31.06fileI suppose I should be all pro-Digium butu meh, use what works!
02:31.14[TK]D-Fenderfugitivo : Thats a fine card.
02:31.20fugitivook, i
02:31.22fugitivoi'll try that
02:33.58CrashHDwhere can I find out what regext= does in the sip.conf
02:34.02CrashHDvoip-info won't show me
02:34.09CrashHDsays I'm not that cool
02:34.11[TK]D-FenderCrashHD : its there
02:34.21filewhat'cha wanna know aboot it
02:34.28CrashHDjust what it does
02:34.32CrashHDI know there is a option
02:34.36CrashHDthat creates an extension
02:34.38[TK]D-FenderCrashHD : its damn near useless....
02:34.38CrashHDin a context
02:35.20CrashHDit's the "damn near" part I'm worried about
02:35.21CrashHD:)
02:35.35fileif you think creatively it has purposes
02:36.23CrashHDalso I was hoping someone could explain what why there are three auth fields for sip devices? when only 2 seem to affect anything?
02:36.42CrashHDuser, auth name, secret
02:36.52CrashHDand how those correspond to the asterisk config
02:39.41znoGnever really looked into the difference between username and auth name
02:39.41[TK]D-Fenderfile : very little use... yeah I can picture one or two, but there are many ways to get around them without this feature...
02:39.59znoGi mean why would someone want a username different to an auth name
02:40.03CrashHDlol, you guys teasing me, or gonna tell me what it does?
02:40.14znoGi'm sympathizing damnit!
02:40.15znoG:)
02:40.29CrashHDznoG: sorry not you, file and D-Fender
02:40.33CrashHDlol
02:40.40CrashHDsympathy welcome
02:40.43CrashHDhah
02:40.47filethere are reasons for the madness.
02:41.04CrashHDbut those reasons aren't documented where I can find lol
02:41.19filethere are also reasons for that madness
02:41.33CrashHDhah
02:41.37[TK]D-FenderCrashHD : its creates a priority 1 exten in the context of your choice "activating" the priority 2+ worth of scripting that you should have waiting for it.  that way you can creat an IVR that has an exten available ONLY when that user is online.
02:41.43CrashHDsmall group of people trying to rule the world
02:41.54[TK]D-FenderCrashHD : clear enough?
02:42.00CrashHDperfect
02:42.13[TK]D-FenderBasically like a way to have it so that no-one can dial your exten if you aren't connected
02:42.21CrashHDfun
02:42.23CrashHDok
02:42.29[TK]D-FenderAlso known as "virtually worthless"
02:42.34CrashHDhah
02:42.39filehaha... virtually..
02:43.12[TK]D-Fenderfile : Yes, and like the platypus, its sole purpose is as FOOD for something higher up the chain...
02:43.24[TK]D-Fenderor perhaps comic releif.
02:43.31CrashHDI vote for comic relief
02:43.49*** join/#asterisk just_a_guy (n=beetle_b@74.136.209.21)
02:43.52[TK]D-Fendernow that SETVAR thing in sip peers.. now THATS cool...
02:44.03filemaybe, maybe not
02:44.04znoGcould be a good feature though
02:44.12znoGinstead of doing a ChanIsAvail, I could use regexten
02:44.33[TK]D-FenderznoG : and how would you check for it?
02:44.45CrashHDhow stable is the trunk guys?
02:44.46znoGcheck for what?
02:44.55[TK]D-FenderznoG : Only way I know is a GOTO, and things get ugly with that line of thinking.
02:45.00fileCrashHD: if you use trunk in even sort of production, I will thwap you
02:45.01[TK]D-FenderznoG : Check that it exists.
02:45.07znoG[TK]D-Fender: with chanisavail?
02:45.11CrashHDfile: ok, good enough answer
02:45.13filethat being said a beta for 1.4 is due to be released in...
02:45.17file8 days
02:45.23[TK]D-FenderznoG : No, using the "regexten" method...
02:45.25CrashHDfile: what about the multi parking lots for call parking?
02:45.38[TK]D-FenderznoG : ChanisAvail is a smart and FLEXIBLE command
02:46.00CrashHDznoG: better be careful...you never know who wrote what here, you may just offend someone
02:46.03just_a_guyHi. Question: First time setting up Asterisk. I installed Asterisk on a machine (which is behind a router with firewall). My IP within the network is 192.168.0.100. Now I want to get X-Lite running from the SAME machine to connect to the Asterisk server. I'm confused about what the sip.conf and X-Lite settings should be. Should I set nat=no because X-Lite is on the same server? What should my host be in sip.conf? Thanks
02:46.20znoG[TK]D-Fender: oh, well, if the user is registered, i guess the exten is available.. so if they press 5, and user 5 is registered, it will call him/her, if not then the priority will not be there and i can make it say whatever at the "i" (invalid) priority.. or did I understand it all wrong?
02:46.32znoGCrashHD: huh?
02:46.44Strom_Cjust_a_guy: you realy shouldn't be running a desktop environment on your asterisk server
02:46.51znoGCrashHD: what did I say that could potentially offend someone?
02:47.04[TK]D-FenderznoG : yes thats what it would do... but why would you not even want the option to be avaiable?  its not "invalid" so much as "not available".....
02:47.05CrashHDznoG: Just joking, *makes a winking expression*
02:47.28CrashHDmy sarcasim doesn't traverse irc very well is all
02:47.30*** join/#asterisk senv (i=pjt@64.6.177.47)
02:47.30just_a_guyStrom_C, It's just for testing. The next time a cheap desktop comes around, I'll buy it and install Asterisk@Home. I just want to play with it for now.
02:47.42Strom_Casterisk@home is even worse :)
02:48.13Strom_Canyway, i wouldnt worry about NAT settings if you're running it on the same machine
02:49.57znoG[TK]D-Fender: if it's not registered, i don't want it to be available .. this is in the "dial another users extension option" in the IVR
02:50.31just_a_guyStrom_C, Well, so I should set nat=no? What should I put as my host? 192.168.0.100 or dynamic? (I'm asking because try what I may, X-Lite always times out while trying to log in)
02:50.50[TK]D-FenderznoG : Thing is if I want to call you and you're not there.... are YOU invalid :)  Thats the statement that it makes.
02:50.57Strom_Chost=dynamic
02:51.00Strom_Cnat=no
02:51.20znoG[TK]D-Fender: well if you want to call me and i'm not registered, the "i" priority should ask you if you want to leave a message...
02:52.10[TK]D-FenderznoG : No, the "i" is for the overall IVR, and it will have NO idea what invalid option you attempted :)  So WHOSE box should it got to, hmm?  Like I said.. WORTHLESS!
02:52.49znoG[TK]D-Fender: if I have the regexten register the extension in the context [myextensions], won't it go to "i" in [myextensions] ?
02:52.50[TK]D-FenderznoG : And end up in "i" because of a typo?  ICK!  Its a downhll trip...
02:52.55just_a_guyStrom_C, OK. Now how about X-Lite? Should I put 192.168.0.100 as the "Domain/Realm", or cable modem's IP? Same question for SIP/Proxy?
02:53.13just_a_guyStrom_C, (And I might as well add that /var/log/asterisk/messages reports everything fine)
02:53.18[TK]D-FenderznoG : And if you INCLUDE it in another (as is the case for most IVR's) scratch that idea!
02:53.29docelm0OI!
02:53.29Strom_Cjust_a_guy: x-lite is supposed to register with the asterisk box
02:53.35Strom_Cso therefore, put the asterisk box's IP
02:53.50[TK]D-FenderznoG : Also, it STILL doesn't know which person you were trying to contact so you couldn't take VM for them anyways
02:53.51*** join/#asterisk gandhijee (n=gandhije@pool-162-84-82-49.culp.east.verizon.net)
02:54.03just_a_guyStrom_C, Right, except that it doesn't work :-( Perhaps I should try another Softphone - no any simple convenient ones for Linux?
02:54.52*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
02:55.10znoG[TK]D-Fender: can't I do VoicemailMain(${EXTEN}) ? actually ${EXTEN} == i wouldn't it...
02:55.11Strom_Cjust_a_guy: or maybe your sip.conf is munged up
02:55.20[TK]D-FenderznoG : BINGO!
02:55.22*** join/#asterisk fugitivo (n=ajf@201.216.246.181)
02:55.23znoG[TK]D-Fender: i'm now convinced, i shall stick to ChanIsAvail
02:55.30fugitivook
02:55.39znoG[TK]D-Fender: :) was worth asking your view on it anyway, otherwise I could still be thinking it would be cool
02:55.42[TK]D-FenderznoG : Congrats!  Hard learned knowledge!
02:56.07[TK]D-FenderznoG : I thought it was cool too... then I blinked and saw the light :D
02:56.26znoG[TK]D-Fender: the concept isn't bad, i think it's just badly implemented.. they could make a channel variable called ${REQEXTEN} that holds the requested extension
02:57.00CrashHDvoip-info should have a version tag for what asterisk version was current when something was posted
02:57.01znoG[TK]D-Fender: so you could run an AGI that checks if the user actually exists and is simply not logged in (within the invalid ext) or if they truly don't exist and you just tell 'em to press the right buttons
02:57.23[TK]D-FenderznoG : thats could be useful I guess... certainly more than regexten...
02:58.47just_a_guyStrom_C, Could be, but my sip.conf is bare minimum...
02:59.00fugitivo[TK]D-Fender: i changed the card
02:59.06fugitivo[TK]D-Fender: and same problem
02:59.26[TK]D-Fenderfugitivo : you need to completely disable the e1000 AND remove the kernel module
02:59.48[TK]D-Fenderfugitivo : its not enough to leave it disconnected from the lan
02:59.55fugitivoi did
02:59.56fugitivowait
03:00.02fugitivonow i have an irq sharing
03:00.06fugitivodamn
03:00.12fugitivoi can't finish with this server :)
03:00.21Strom_Cjust_a_guy: pastebin the sip.cong
03:00.22Strom_Cer
03:00.26Strom_Csip.conf
03:00.28Strom_C~pb
03:00.30jbotpb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
03:02.11just_a_guyStrom_C, [general]
03:02.11just_a_guycontext=default                 ; Default context for incoming calls
03:02.11just_a_guybindport=5080                   ; UDP Port to bind to (SIP standard port is 5060)
03:02.11just_a_guybindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
03:02.11just_a_guysrvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
03:02.12just_a_guy[myxtenid]
03:02.14just_a_guytype=friend
03:02.16just_a_guysecret=welcome
03:02.18just_a_guyqualify=yes
03:02.20just_a_guynat=no
03:02.22just_a_guyhost=dynamic
03:02.24just_a_guycanreinvite=no
03:02.25CrashHDpastebin is your friend
03:02.26just_a_guycontext=default
03:02.33CrashHDhttp://www.pastebin.com/
03:03.29Strom_Cjust_a_guy: I said pastebin
03:03.43Strom_C~pb
03:03.44jbotpb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
03:04.48just_a_guyStrom_C, Apologies, didn't know what pastebin was: http://pastebin.com/734442
03:05.20CrashHDjust_a_guy: no worries mate, gotta start somewhere
03:05.38fugitivois it ok if now i have irq sharing to compile APIC support?
03:06.04*** join/#asterisk Gabriel25 (n=whyban@user-12ld5f7.cable.mindspring.com)
03:06.14[TK]D-Fenderfugitivo : Sure, why not....
03:06.50fugitivook
03:09.05*** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca)
03:09.54senvi am haing wierd problems with my tdm400p. incoming lines connect then hangup
03:10.12senvis this a common problem?
03:10.47*** join/#asterisk bjohnson (n=bjohnson@i216-58-51-95.cybersurf.com)
03:10.48senvkeeps getting zap at incoming failed
03:11.12senvi can dial out on the lines fine
03:12.12Strom_Csenv: what context do the fxo channels live in, and do you have an s extension in that context in nyour extensions.conf?
03:12.21fugitivosenv: sounds like a context problem
03:12.23*** join/#asterisk coppice (n=chatzill@120.195.17.210.dyn.pacific.net.hk)
03:13.24[TK]D-Fendersenv : pastebin the call attempt at CLI on verbose 10
03:13.26[TK]D-Fender~pb
03:13.27jboti heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca
03:13.49*** join/#asterisk ManxPower (n=ewieling@69-2-85-41.wan.networktel.net)
03:16.16senvok it is http://pastebin.com/734454
03:17.32Gabriel25can someone can help me
03:17.44fugitivosenv: pastebin your extensions.conf ONLY if your
03:17.50Gabriel25I what somene to try to connect from an external ip to my asterisk box
03:17.52fugitivoyou're not using asterisk@home
03:17.59fugitivo^^^ senv
03:18.16senvi am using asterisk@home
03:18.25fugitivolol
03:18.35senvfreepbx
03:18.43Strom_Cgo to #freepbx
03:18.48senvlol. o k
03:18.51senvthanks :)
03:19.06*** part/#asterisk senv (i=pjt@64.6.177.47)
03:19.44fugitivo[TK]D-Fender: still the same problem :(
03:22.04[TK]D-Fenderfugitivo : Did you disable the E1000 in the bios?
03:22.41[TK]D-Fenderfugitivo : And remove the add-in one?  And kill the kernel module?
03:22.42*** join/#asterisk hacked`` (n=lol@modemcable226.130-37-24.mc.videotron.ca)
03:22.53hacked``guys
03:23.01hacked``lets say i wanted to set up so that whoever calls me, they're prompted with a message, like "to reach customer service, press 1", etc
03:23.06hacked``and it would route the call to the right person
03:23.08hacked``what do i need for that
03:23.17fugitivo[TK]D-Fender: disabled from the bios and from the kernel
03:23.18znoGAsterisk !!
03:23.24fugitivo[TK]D-Fender: and rebooted
03:23.28hacked``znog, ya but i mean in terms of hardware
03:23.43znoGwell that depends on many things
03:23.57InfraRedhacked``: depends on how you want to connect to the pstn
03:24.01fugitivo[TK]D-Fender: i'm using a 3com right now
03:24.01znoGlike how many incoming lines you have/will have, how many extensions, etc... i'm willing to bet you haven't even touched voip-info.org :)
03:24.09hacked``infrared, what are my choices here
03:24.18InfraRedwe're not sales people
03:24.19hacked``znog, correct
03:24.25InfraReddo your onw research
03:24.27InfraRedown
03:24.31hacked``infrared, this is part of my own research
03:24.42InfraRedvoip-info.org then
03:24.47znoGi'm willing to bet this was your first option in your "research" :)
03:25.03hacked``actually i read the asterisk wiki, and its all gibberish to me
03:25.03InfraRedreasearch by asking someone else
03:25.09znoGif you want people to willingly help you, it's always good to show you did a little research on your own first
03:25.22[TK]D-Fenderfugitivo : Hmmmmm...
03:25.24znoGit's easy to say "i want X and Y to do Z, what do I do?"
03:25.34hacked``im not asking you how to set up asterisk, im asking what i need, in general
03:25.55[TK]D-Fenderhacked`` : Depends.  What kind of lines are you intending on using?
03:26.41InfraRedhe plans to spam people with phonecalls!
03:26.42fugitivo[TK]D-Fender: a big hmmmmmmmmm
03:26.46hacked``all i want to do is buy a couple ip phones, set them up on my network, have 1 incoming pstn line, and have a prompted to redirect calls to specific phones
03:27.04InfraRedhow is that incoming line presented?
03:27.13InfraRedanalogue/isdn
03:27.15ManxPowerThese business trips really put the miles on the car
03:27.20hacked``its just a regular analog line
03:27.34InfraRedget a digium fxo card then
03:27.48hacked``what about an fxs card
03:27.54InfraReda server and couple of ip phones
03:28.00InfraRedwhat about it
03:28.02InfraRedthey're nice
03:28.07hacked``why dont i need one of those
03:28.12InfraRedrtfm its all described there
03:28.15hacked``doesnt make sense that i would need 1 fxo but not a fxs
03:28.20ManxPowerWell, a Digium TDM400P w/FXO or a SIPura SPA-3000, or a Clone X100P card, or, or, or
03:28.22InfraRedr t f m
03:28.33InfraRedesp the fxo vs fxs part
03:28.38hacked``k
03:28.39ManxPower~fxofxs
03:28.40jbotit has been said that fxofxs is An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this.  An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this.
03:28.42InfraRedvoip-info
03:28.45[TK]D-Fenderhacked`` : any of what ManxPower jsut suggested + the A200.
03:29.02ManxPower[TK]D-Fender: heritic
03:29.12InfraRedA200 ?
03:29.14[TK]D-FenderManxPower: zealot
03:29.20[TK]D-Fender:D
03:29.26ManxPower[TK]D-Fender: 8-)
03:29.26InfraRedwhats an A200
03:29.29InfraRed~A200
03:29.38ManxPower[TK]D-Fender, I'm planning on going with Sangoma for my next project
03:29.53[TK]D-FenderManxPower : hehe, fine!
03:30.00[TK]D-FenderManxPower: hypocrit
03:30.00[TK]D-Fender:D
03:30.01ManxPowerInfraRed, Sangoma.  They make cards that work with Asterisk using their own drivers to emulate the Digium interface.
03:30.07InfraRedah nice
03:30.14ManxPower[TK]D-Fender: Kids, don't do drugs!
03:30.31InfraRedcisco phones are overrated
03:30.34hacked``but so can i buy any ip phone, regardless, cause i was thinking of buying those cisco phones like on 24 so i can have that ringtone
03:30.38[TK]D-FenderManxPower : Stay in milk! Don't do school!  Drink your drugs!
03:30.41InfraRedthey;re PITA to install, PITA to configure
03:30.56[TK]D-FenderInfraRed : Not so must overrated as overpriced.
03:31.11*** join/#asterisk voipaster (n=25x8supp@203.167.120.9)
03:31.20[TK]D-Fenderhacked`` : You can get the ringtone working on ALL SORTS of phones...
03:31.25ManxPowerhacked``, Get Polycom.
03:31.26fugitivoi think i'm not going to buy digium cards anymore
03:31.36voipasterhi
03:31.42[TK]D-Fenderhacked`` : Linksys SPA9xx have it as an option by default as they are made by Cisco.
03:31.44InfraRedhacked``: cisco is more hassle than its worth
03:31.48ManxPowerCiscos don't come with SIP firmware (extra cost) and don't come with a power supply (extra costs)
03:31.55[TK]D-Fenderhacked`` : Yup, a huge thumbs up for Polycom.
03:32.04voipasterim new on asterisk, im installing my digium card te110p
03:32.32voipastercan someone give me a step by step on installing and configuring it pls
03:32.55InfraRedvoipaster: www.voip-info.org
03:32.57voipasterok thanks
03:32.58InfraRed~docs
03:33.00jbotmethinks docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
03:33.05InfraRedits all there
03:33.07InfraRed^^^
03:33.21ManxPower~thebook
03:33.28InfraReddont expect this to take few hours
03:33.31InfraRedplan days
03:34.02ManxPowerIt was hard for us to learn, it should be hard for you to learn.
03:34.45Strom_Chire a consultant to do it for you
03:34.53Strom_CI'm a bargain at only $125 per hour
03:36.04[TK]D-FenderWas easy for me.. but I'm an adept :)
03:36.06ManxPowerI'll do it for $120/hr as long as I can insult you too.
03:36.22QwellI'll do it for $115/hr as long as I can insult ManxPower too.
03:36.34JunK-Ymouhahah
03:36.35ManxPower"Here's my bill.  You're ugly and your mother dresses you funny."
03:36.47[TK]D-FenderQwell : Heck, I do that for FREE!  Am I not truely AWESOME?
03:37.37Strom_CFor $125 per hour I throw in insults to the idiot who set up the system that I'm rebuilding and setting up correctly :)
03:37.55QwellStrom_C: you win
03:38.02InfraRedcheap bastards
03:38.20*** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)
03:39.29[TK]D-Fendervoipaster : Before even talking about your TE110P, how much have you worked with *?
03:40.02InfraRed[TK]D-Fender: he just said he was new
03:40.16gandhijeewhoever was thinkin about polycoms, don't get them if u want to use the microbrowser to actually do shit
03:40.48[TK]D-Fendergandhijee : Have you talked to their techs about it?
03:40.49ManxPowergandhijee, only the 600 has a microbrowser anyway.
03:41.01gandhijeeyah i know
03:41.07ManxPowerif you want a microbrowser then go with cisco and pay the extra money
03:41.28gandhijee[TK]D-Fender: i emailed them yesterday from that knowledge page they had, tryin to get some documentation on it
03:41.59gandhijeewe'll see what happens
03:42.36[TK]D-Fendergandhijee : well the double-post issue should be traceable
03:42.50gandhijeei really don't think thats in my code
03:43.00CrashHDanyone have an updated website about running multiple asterisk installations
03:43.15gandhijeei think it has to do something with the way the polys handle forms
03:43.16[TK]D-Fendergandhijee : I didn't think so either.... so they shoudl be able to figure it out.
03:43.18Gabriel25guys I have a problem .... when I try too add an external SIP phone dosen`t work
03:43.27[TK]D-Fendergandhijee : did you try to break it down like I suggested?
03:43.29Gabriel25what can be the problem ?
03:43.31gandhijeeyeah
03:43.48ManxPowerGabriel25, it could be a billion differnet things.
03:43.52gandhijeefor the XML error thing, i think the page was just too large
03:44.07gandhijeetoo much code for it to handle, which seems kind of retarded
03:44.13[TK]D-FenderManxPower : Especially since he's using AMP....
03:44.17ManxPowerassuming Asterisk is on a public IP address, all you should need is nat=yes
03:44.20Gabriel25ok my linux box have 2 network cards one is LAN and one WLAN from LAN is working fine from WLAN dosen`t work
03:44.23gandhijeebut then again i should be breakin some of the stuff down in to modules/functions
03:44.26hacked``why should it be hard
03:44.33ManxPower[TK]D-Fender, Ah, then it could be a quadrillion different things
03:44.59[TK]D-FenderManxPower : I believe the appropriate term is "arbitrarily large number" ;)
03:45.02Gabriel25ManxPower where I can put nat=yes?
03:45.13ManxPowerGabriel25, Is the phone behind NAT?
03:45.18Gabriel25in the server config files or on the SIp soft phone ?
03:45.32Gabriel25I don`t know how to do this
03:45.36ManxPowerGabriel25, all NAT for clients is in sip.conf.
03:45.41ManxPowerGabriel25, read The Book
03:45.49gandhijeeGabriel25: so your softphone doesn't work in the WLAN?
03:45.51ManxPower~thebook
03:46.01jake1932jbot is dead
03:46.17pjchilds~slap jbot
03:46.18jbotACTION slaps jbot, keep your grubby fingers to yourself!
03:46.54jake1932or not
03:47.51Gabriel25Thank you ManxPower
03:48.27ManxPowerIf you set nat in asterisk and nat in the SIP client, well two nats don't make it right.
03:49.36pjchildsanyone use a session-border-controller gateway in-the-real-world ?
03:50.46docelm0yes
03:50.47docelm0why?
03:54.58pjchildsdocelm0, just what type of SBCs people were using, and what they thought of them...
03:54.58Gabriel25ManxPower thank you is working now I had to change from nat=never
03:55.09Gabriel25to nat=yes
03:55.12Gabriel25and is working !
03:55.15docelm0MERA can suck my D!CK..   I HATE IT!
03:55.22Gabriel25thank you so so much
03:55.34docelm0its one of the worst..  but MVTS II is supposed to be better
03:56.39docelm0actually the worst would be PortaOne..   They suck big time..  they call themselves a SBC but they are not even close
03:57.20docelm0I messed with nextone..  it wasnt too bad..
03:57.38pjchildsamcepacket?
04:00.18[TK]D-Fenderpjchilds : I wouldn't trust a company starting with acme.... I remember this one poor coyote...
04:00.28Qwell[TK]D-Fender: That was a different acme
04:00.52jake1932it was ACME not acme
04:01.12docelm0The only guys who use acme I know of are GX and XO
04:01.31fugitivothis is a mess
04:01.41fugitivo99.902344% 99.902344% 99.890137% 99.743652% 99.536133% 99.902344% 99.902344% 99.890137%
04:02.42pjchildsfugitivo, time to get a sagnoma ? :)
04:03.14znoGfugitivo: i get 99.70% on a FXO card... one fax out of 10 make it
04:03.26znoGi have to make it go on IRQ 9 which is not the easiest thing to do
04:03.29znoGon this mobo anyway
04:03.30fugitivotime to escape to another country
04:07.58fugitivodigium support is great
04:08.19fugitivowhat is the response time for emails? 1 week?
04:08.23CrashHDzttest should be 100% all the time right?
04:08.38fugitivoit should
04:08.44znoGwell i get 99.95% or something which I consider quite cood
04:08.45znoGgood
04:08.49fugitivobetween 99.98 and 100
04:08.49JunK-YCrashHD: not necesseraly
04:08.51znoGoccasionally it hits 100% but rarely
04:09.08CrashHDahh ok
04:09.40CrashHDI have an e1000 in that box and it is still 99.8+
04:15.36gandhijeeafter i loaded irqbalance i got better results on my box
04:15.45asterboyAnyone setup a Bogen Lucent paging system?
04:15.49gandhijeebut its a hodge podge of hardware
04:15.49*** join/#asterisk meesterfox (n=M_fox@71.224.224.168)
04:16.03asterboyPlaying with a LUPCMALL
04:16.32asterboyHas TIM,CPU,TBM and ZPM modules
04:16.34meesterfoxAnyone have any experience with the asterisk wakeup call annoy script? or one of it's variants...
04:17.04asterboyJust want to know if there is a way to hookup a speaker without an AMP.
04:17.22asterboyManual says you can use 70V
04:17.52gandhijeehey can someone take a look at this error
04:17.52gandhijeehttp://pastebin.com/734494
04:18.03asterboyNot sure if I should use an FXO or FXS module to connect it.
04:18.08asterboywants loopstart
04:18.10gandhijeeits giving me some crap about ODBC and bad SQL
04:18.19asterboyso I'm thinking FXO
04:19.26asterboyotherwise, I'll just hookup something to the sound port of the * box.
04:19.34asterboyproblably the easiest
04:23.36*** join/#asterisk rstrit (n=rstrit@204.238.218.130)
04:28.37*** join/#asterisk supjigatr (n=syslod@152.53.16.10)
04:28.47supjigatrHi.
04:29.51*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
04:31.34*** join/#asterisk sudhir492 (n=sudhir@pool-71-114-99-2.washdc.dsl-w.verizon.net)
04:31.39sudhir492Hi all
04:31.54sudhir492anyone using spandsp and rxfax heres?
04:33.26supjigatrYep
04:33.31*** join/#asterisk sudhir492 (n=sudhir@pool-71-114-99-2.washdc.dsl-w.verizon.net)
04:33.42supjigatrWell I was till my sangoma driver stop loading after a reboot.
04:33.47sudhir492Awfully quiet in here
04:33.49mitchelocisn't spandsp replaced by asterfax now?
04:33.50supjigatrYep
04:34.01supjigatrhuh?
04:34.01sudhir492asterfax, hmm
04:34.07supjigatrspandsp is asterfax
04:34.14sudhir492let me check. I thought that asterfax uses spandsp
04:34.21supjigatrIt does!
04:34.30*** join/#asterisk ghost99 (n=neville@222-153-178-14.jetstream.xtra.co.nz)
04:34.54*** join/#asterisk the_real_JasonF (n=jfrisch@60.32.160.80)
04:35.02the_real_JasonFHello all
04:35.06supjigatrHi
04:35.08fugitivowhat is asterfax?
04:35.17mitcheloctry google?
04:35.33the_real_JasonFI am trying to redirect incoming calls to an outside number, but unless I pick up the call it is failing
04:35.44supjigatrBefore I kick this machine, does anyone have a sangoma card a104d working?
04:35.46the_real_JasonFis there anyway to answer and call and redirect it automatically...?
04:35.52supjigatrYea
04:36.02supjigatrdial
04:36.17the_real_JasonFdial doesn't pick up the call..
04:36.26[TK]D-Fendersupjigatr : I do, what of it?
04:36.43sudhir492yes
04:36.54sudhir492I have sangoma A104 working
04:36.56the_real_JasonFmaybe I should just Answer, then dial with r
04:36.59sudhir492works great
04:38.13the_real_JasonFhehe, answered my own question..
04:38.27the_real_JasonFbut I wonder why I need to answer :-(
04:38.49supjigatrI can't seem to keep it working.
04:39.00sudhir492the_real_JasonF: Try something like
04:39.00sudhir492exten => 7034441234,1,Dial(Zap/g1/12124567890,25)
04:39.35meesterfoxanyone have any experience with the wake up call script?
04:40.31meesterfoxmine works fine, except when you hangup it doesn't call you back or anything.
04:40.32supjigatrD-Fender: Are u using the ec and hdlc on the card?
04:41.41supjigatrI suspect its UDEV problem but I can't seem to figure it out.
04:41.44sudhir492Unfortunately, I do not remember what I configured the card with. But it works great so far
04:41.59sudhir492just follow Sangoma's instructions
04:42.20supjigatrI followed them.  I'm running slackware.
04:42.32supjigatrAre you using UDEV?
04:42.37sudhir492hmm. I am running FC3
04:42.38supjigatrlinux 2.6.x?
04:42.48[TK]D-Fendersupjigatr : you, the works.
04:43.35supjigatrI want it all too but I had it working once but after reboot it fails.
04:44.42the_real_JasonF<supjigatr> < I actually tried just a dial, but it gets rejected
04:44.43supjigatrRight now I get this wanpipe FATAL: Error inserting af_wanpipe
04:45.23the_real_JasonFthe only difference I can see is that the IP from IP changes to the asterisk server ( or in internal IP if I redirect manually)
04:45.51the_real_JasonFbut if i Answer it in the dialplan, I figure it means the callers starts getting charged..
04:46.32the_real_JasonFie. dials out via sip, no POTS
04:46.37sudhir492yes, Linux 2.6
04:49.06[TK]D-Fendersupjigatr : where do you get that error?
04:49.52supjigatrwanrouter start
04:50.45*** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net)
04:50.50[TK]D-Fendersounds like amke a permissions thing...
04:50.56[TK]D-Fendermaybe*
04:51.09supjigatrWhere?
04:52.01*** join/#asterisk Crshman (n=chatzill@hacienda-heights-cuda2-68-71-5-62.lmdaca.adelphia.net)
04:52.02[TK]D-Fendersupjigatr : not entirely sure..
04:52.09Crshmanif i put "follow me" on an extension will that extension ring? or only the numbers in the "follow me" list?
04:53.10[TK]D-FenderCrshman : What is this magical "follow me" of which you speak?
04:53.29Crshmanooooo that's right it's only in FreePBX, oops
04:53.38InfraRedstalker mode pbx
04:53.39Crshmanok wrong channel then sorry
04:53.51*** join/#asterisk angler- (n=angler@pdpc/sponsor/digium/angler)
04:54.22[TK]D-FenderCrshman : No, only the version YOU are talking about...
04:54.29*** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz)
04:54.39Crshman[TK]D-Fender: ? i don't understand
04:54.54iceyphey guys, I have 2 729 codecs from digium, when i make a call from my cisco phone and try and conference another person in, i get this error: May 24 16:54:05 WARNING[46448]: codec_g729.c:259 lintog729_framein: Out of G.729 Encoder Licenses!
04:55.37[TK]D-FenderOk, I'm fried.. back tomorrow peeps....
04:56.26iceypshould sip.conf not go in order of codecs , i.e. disallow=all, then allow g729, then if no licenses left drop down to next option ulaw?
04:56.27JunK-Yiceyp: show g729
04:56.51*** join/#asterisk jero (n=jero@modemcable235.87-82-70.mc.videotron.ca)
04:57.09iceypthe fact is if I have any other codec in the sip.conf under the cisco phone, the it will use it, i.e. no matter where in the context I put allow=ulaw, it uses that over 729
04:57.37Strom_Ciceyp: one license to talk to the phone, one license to talk to the outside world, and then oops, out of licenses!
04:57.56iceypstoffell then it should go to ulaw or something?
04:58.14iceypit uses 2 licenses even when i call voicemail or meetme
04:58.23Strom_Cuse ulaw locally
04:58.35iceypmy pabx is remote to me
04:58.45iceypso i use 729 to the pabx
04:59.03iceypbut if there is no codecs it should drop to ulaw as next option, rather than not let me make a call at all
04:59.16SwKyou dont need 2 licenses of G729 to talk thru the PBX
04:59.27jeroanyone experienced issues with voip without qos on a 100mbps lan ?
04:59.28SwKyou only need 1 license for each transcoding session
04:59.40SwKjero: only when the LAN is heavily loaded
05:00.07jeroswk: what is heavily? 1 transfer between 2 hosts ?
05:00.16jeroor many many ones
05:00.48SwK1 transfer between 2 hosts can congest the network... it really depends on what type of hub/switch you have and what the hosts are
05:01.20SwKthen again some networks take many transfers between many hosts to get truely congested
05:01.33SwKit all depends on the hosts and related network hardware
05:02.05iceypstupid stupid codecs
05:02.21Strom_Cg729 blows anyway
05:02.30litagecan a linux box's hostname begin with an underscore?
05:02.51Qwelllitage: I don't see why not
05:02.57SwKi'm sure it could start with whatever printable ascii you wanted it to
05:02.59iceypif i have no codecs specified in sip.conf [general] section, will this stop all calls?
05:03.08jerothanks swk
05:03.12Snake-EyesIs there any way to stop asterisk from complaining about using inband dtmf when using g729 codec? Inband works very well with my current setup where rfc2833 doesnt.
05:03.26Strom_CSnake-Eyes: you cant use inband dtmf on g729
05:03.38Strom_Cat least not with any hope of reliability
05:03.54litagethanks Qwell
05:03.54SwKDTMF inband with g729 might work but its far from usable in most situations
05:03.55Strom_Cthe codec fucks up the sine waves
05:04.17Snake-Eyeswell i am and seems work fine so far.
05:04.29SwKwhat rfc2833 issue are you having?
05:04.50Strom_CSnake-Eyes: if you want inband dtmf, use ulaw
05:04.56SwKor alaw
05:04.59Strom_Cotherwise, use info or rfc2833
05:05.00Snake-Eyesis it posiable that asterisk is complaining that the one channel is g729, but channel going out isnt
05:05.36iceypdoubt it
05:05.43Snake-Eyeseg channel to end pt is ulaw and channel coming back is g729 ?
05:05.58iceypseems that asterisk is using the codecs inlisted in my [general] rather than trying to use the codec peer specific
05:06.27Snake-Eyesevery time i setup rfc2833 nothing seems to registry on the other side eg navigating pbx menu system
05:06.36SwKiceyp: codecs listed in general set the overall codecs and if its disabled there its disabled everywhere
05:07.01SwKwhat ATA are you using?
05:07.18SwKwhat ATA with the 2833 issue that is?
05:07.32Snake-Eyesive tried it on gxp-2000 and spa941
05:07.37iceypi'm using a cisco 7912, [general] disables 729 but my cisco 7940 context enables 729
05:07.41iceypand i can call with 729
05:08.19iceypmaybe i broke something with peer/user/friend
05:08.21Snake-Eyeshmm, i have feeling the people doing line/pstn termination might be one reason
05:08.45Snake-Eyesi have no codecs allowed or disallowed under [general]
05:12.06iceypweird, now it uses ulaw even though i've disallowed everything
05:12.35iceypMay 24 17:12:27 NOTICE[50174]: chan_sip.c:3646 process_sdp: No compatible codecs!
05:12.43iceypand disallow=all in [general]
05:12.56iceyphowever on the specific user i've allowed g729 & ulaw
05:14.18Snake-Eyesi have seen it where ulaw, alaw and g729 are allowed, which will result in either g729 being used every time  or g729 being ignored completely. This is supposed be fixed when asterisk 1.4 comes out thou
05:14.51iceypmmmm ok
05:14.55iceypso its known
05:15.17iceypbut how come if i disallow=all in [general] it doesnt use the allowed ones in the users context?
05:15.18Snake-Eyesyea
05:15.47iceypit didnt use to act that way
05:16.06Snake-Eyesdont know
05:16.21iceypmaybe its the type=friend vs type=peer thing now
05:16.46Snake-Eyesdid you have disallow=all under users?
05:16.58iceypyes
05:17.08iceypand then allow=g729 allow=gsm allow=ulaw
05:17.14iceypin that order
05:18.05iceypwhen i set type=friend then it uses the codecs below that user
05:18.26Snake-Eyesim understanding is the local context (peer, friend, user) overides the general context
05:32.13*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
05:40.34Flautohow call parking works?
05:40.42Flautois there anyone willing to help
05:41.51Strom_Cyou give the call to the valet attendant along with a $5 tip and tell him to take extra-special care of your baby
05:42.39*** part/#asterisk the_real_JasonF (n=jfrisch@60.32.160.80)
05:50.31Crshmanis there a limit as to what extensions i can use?
05:51.03*** join/#asterisk Kis (i=vlad@p5080FD06.dip.t-dialin.net)
05:51.08Strom_Cwhat do you mean, Crshman?
05:51.18*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
05:51.24Crshmanlike i can use 1xx 2xx 3xx 4xx with no limitations?
05:51.27*** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com)
05:51.35Strom_Cwhat do you mean "limitations"
05:51.43Crshmanare any extensions blocked out?
05:51.53drrayI don't think any extensions are blocked off by default in extensions.conf
05:51.56Strom_Cwhat do you mean "blocked out"?
05:52.00drrayfeatures maybe
05:52.11Strom_Cyou can assign anything you want anywhere you want
05:52.20drraythe * features might supercede
05:52.30Strom_Cwell sure, vertical service codes
05:52.35Crshmano
05:52.43Crshmanwhere is a listing of those?
05:52.48Strom_Cbut thats why you look at www.nanpa.com to  find out :)
05:53.04drrayfeature.conf
05:53.06Strom_Ci believe you can assign your own vertical service codes in the *95-*99 range
05:53.10drrayer, features.conf
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05:53.31Crshmanexcelent thnx
05:53.31Strom_CCrshman: leave vertical service codes to nanpa specs
05:53.37Strom_Cwww.nanpa.com
05:54.09Strom_Conly conflict with the existing numbering plan if you know EXACTLY what you're doing
05:54.38Crshmanno i'm trying not to =)
05:54.43just_a_guyUpon loading asterisk, I get the following error: [codec_speex.so]Ouch ... error while writing audio data: : Broken pipe
05:54.52just_a_guyIdeas?
05:54.56drrayCrshman - I use #434 for an extension
05:55.02drrayincluding the # sign
05:55.27Crshmanooooooooooo so not just the 4xx number? you add a #?
05:55.39drrayit can be any number you want
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05:56.02Crshmanok
05:56.39drrayit could be ABCD if you had a phone set that supported it
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06:02.30Strom_Cdon't use # to start an extension
06:02.41Strom_C# indicates completion of dialing
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06:04.32Crshmanhow much processor does software echo cancellation use?
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06:04.55beetle_b2Sorry - I lost the connection last time. Question: Running asterisk gives me the error: [codec_speex.so]Ouch ... error while writing audio data: : Broken pipe
06:07.13Strom_Canyone here have experience configuring sangoma cards on PRI circuits?
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06:25.15Crshmani'm having a great issue with echo on my setup.....how can i fix this?
06:26.32Strom_CCrshman: what kind of lines are you using?
06:26.45CrshmanIAX out and SIP in
06:27.03angler-Crshman, not much u can do in all voip
06:27.23Crshmanso there is no fix for the echo problem?
06:27.35Strom_CCrshman: what kind of telephone set are you using?
06:28.02Crshmani'm using a softphone to dial out to a cell phone and i get echo
06:28.24Strom_CCrshman: do you have a headset for the softphone, or are you using a microphone and speakers?
06:28.47Crshmanmic and speakers, i have enabled the "echo cancellation" for mic and speakers in the softphone app...
06:29.01Strom_Cthere's a reason quality speakerphones cost hundreds of dollars
06:29.12Strom_Cthe echo is coming from your poor setup
06:29.17Strom_Cget a headset or a real telephone
06:29.19angler-id say so too
06:29.30Crshmano ok i thought it was the actual lines
06:29.33Crshmani'll try it
06:29.50Strom_Cso, any sangoma zealots out there willing to help with a PRI config issue?
06:30.45angler-maybe if it was digium pri card
06:30.56drrayisn't it the same damn thing?
06:30.56Strom_Cyeah, I know, I much prefer the digium cards
06:31.04*** join/#asterisk lorinc (n=ang@caracas-4338.adsl.interware.hu)
06:31.17Strom_Cbut I've been brought in to fix this fucked up asterisk install, and the previous installer used a sangoma card
06:31.42Strom_Cdrray: no, the sangoma cards patch zaptel and have sixty-five extra ass-backwards config utilities
06:32.14angler-drray, I won't touch sangoma
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06:32.37Strom_Cneither will I when I'm specing out the hardware - this is ridiculous
06:32.45angler-hehe
06:33.07drraythat 8 port sangoma card would solve a few of my problems
06:33.19Strom_CI'm surprised - usually you start bagging on sangoma and some zealot pops out of nowhere to tell you how wonderful sangoma is
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06:33.39Strom_Cno such luck tonigt, I guess
06:33.51achandra_hello, I posted earlier about using the openser module dispatcher to load balance...which i have working...however the failover part...can someone help with it?... when asterisk box fails?...here is the doc - http://openser.org/docs/modules/1.1.x/dispatcher.html
06:34.12pjchilds(zealot mode) ooh... sangoma is wonderful...
06:34.39achandra_the doc does have the use of flags to deal with it...
06:34.50achandra_but im unclear on the nomenclature..
06:35.31Crshmancool thnx folks i got it fixed using a better headset echo is nearly gone
06:35.57stephane_jour
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06:39.09supjigatrAnyone recommend disto that plays nice with sangoma a104d?
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06:40.27FuriousGeorgehey all
06:40.41nfinetinhi
06:40.50FuriousGeorgejust wondering, i got a server im building thats gonna have 2 tdm400ps
06:40.50supjigatrHi
06:40.59FuriousGeorgeso irq sharing is a concern
06:41.11Strom_Cuse a tdm2400 :)
06:41.15FuriousGeorgeshould i get a pci or pci-e gpu?
06:41.44FuriousGeorgedoes it matter?
06:43.04FuriousGeorgei mean, it is higher end tyan mb, once i get it to post it should be able to assign separate irqs to 3 pci devices, right?
06:43.35supjigatrYou will likely have problems.
06:43.43FuriousGeorgethen again i got those little tiny pci slots.  i guess those are pci 1x lemme get that manual
06:44.59Strom_Chow charming!  when you unload the sangoma module, the system grinds to a screeching halt
06:45.04Strom_Cthat's just fabulous
06:45.48FuriousGeorgeok so i got 4 pci slots, two will be used for tdms, then i got 2 pci-e X1 and 1 pci-e X16
06:45.50drrayright now I'm using a go varion.com tor2 clone, and am thinking about getting the new Sangoma 8 t1 port card
06:45.54pjchildsachandra, I would assume you would use ds_select_dst() in route{} and then t_relay() [stateful] -- then in failure_route[x] call ds_next_dst() before t_relay()
06:46.50FuriousGeorgedoes it make sense that my pci-e slots are less likely to share an irq with the other pci slots?
06:48.47pjchildsachandra, but the docs aren't very specific, like 'how do you know if the list is empty', and 'how do you re-enable a destination you have marked dead with 'ds_mark_dst()' ...
06:50.08*** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr)
06:50.17tparcinahi group!
06:50.30nfinetinhi
06:50.46Strom_Chi!
06:50.54nfinetinis there a way to use an other prompt language ? like french or german ?
06:51.06angler-nfinetin, yup
06:51.36FuriousGeorgeit seems that pci-e may be on an entirely separate bus and wont share irqs at all with pci.  so i will likely not have problems
06:51.37tparcinaanybody uses asterisk-stat
06:51.44angler-nfinetin, Use set for the language variable to change it to a different sounds directory
06:52.13tparcinai have problem conecting to mysql database. asterisk-stat can't connect and show data from mysql
06:52.19nfinetinwhere can i change that variable i mean witch confile ?
06:52.51tparcinaset(language(de))
06:53.10tparcinanfinetin, something like that - check on voip-info
06:53.38tparcinalook for application - set
06:54.17tparcinaso, anybody knows how to check why asterisk-stat doesn't connect to mysql database?
06:54.31nfinetinok thx, an other thing is that i cannot figure out how to run rxfax and txfax according to docs
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06:55.31tparcinanever used rxfax and txfax, sorry
06:56.04angler-nfinetin, those apps are apart of spandsp, the docs on their site seem pretty straight forward
06:56.35nfinetinis there something else that i could use for faxes on *
06:56.49angler-depends
06:56.52nfinetinrunnning with freepbx
06:57.25angler-well i would stick with spandsp
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06:58.19ZeeekManxPower ?
06:58.21CrashHDif a call is sent to a queue with QUEUE() and no members are avilable and joinempty = strict?
06:58.29CrashHDwhat happens to the call?
06:59.44Strom_CCrashHD: dead hookers
06:59.58angler-lol
07:00.00angler-what?
07:00.27CrashHDlol
07:00.29CrashHDoff the wall
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07:07.28Crshmanwhat is a good TTS program?
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07:07.45satlan32good morning..
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07:08.05satlan32good morning..
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07:19.37fugitivohi
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07:21.11satlan32hi
07:21.22satlan32anyone used audiocodes mp 104 fxo?
07:21.47Zeeekhi
07:21.49satlan32i was googeling for a day but all i can find is where to buy one and not how to configure it to work with asterisk
07:23.19*** join/#asterisk Bart` (n=yann@gw2.overlap.fr)
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07:23.36Assidheya
07:25.19Assidwhats the best way to connect a legacy pbx into voip
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07:26.13satlan32voip gateway
07:26.30satlan32such as digium cards, audiocodes media gw
07:27.09Assidwell..  i was thinking of digium cards.. but.. i keep getting mixed up between fxo and fxs
07:27.16Assidfxo is where you dont procduce dialtone right
07:28.44Assidhey the linksys pap2 would do as well right ?
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07:53.06nfinetinhye again, i've figure out how to install sapndsp it was a patching issue
07:53.11nfinetinthx for your help
07:53.54nfinetinwhy do i get line twice times in the feature vcore admin modules
07:54.29angler-thats probably a question for #freepbx
07:55.23nfinetinok sorry wrong window
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08:02.48bartlebeecould anyone have cisco ip phone 7960 firmware ver 6.x they're willing to share please?
08:02.54bartlebeecould = would
08:03.55x86try #cisco
08:06.07bartlebeek, waiting on #cisco to be directed to a url and contact your nearest helpful cisco dealer
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08:11.40MrChimpyhey *ers
08:13.08bartlebeeif we tell you how will you let MrChimpy know?
08:19.05stephane_re
08:19.48*** join/#asterisk Tusker (n=tusker@203.117.94.152)
08:19.55bartlebeeMrChimpy :)
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08:22.19Tuskerheya guys... if I have an incoming call, and I have a context set for it, how do I configure for that incoming call ? ie, I have [sip-provider] in sip.conf, with context=sip-provider-context, when a call comes in, it says it is 'Looking for blah-user in sip-provider-context'.  Where do I define that user? :)
08:22.45Tuskeroh... and asterisk is responding 'SIP/2.0 404 Not Found'
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08:31.59MrChimpywhy have one interface when you can have three different ones?
08:32.12angler-Tusker, should be looking for an extension in the context
08:32.37Tuskerangler-: ahh i see... rather than s, just specify blah-user... thanks!
08:33.43Tuskerok, another question... 'Their Codec Capability:   271' < how do I know what formats the 271 mask is ?
08:34.31angler-look at sip debug, it will tell what each codec is
08:35.32x86Tusker: 'show codecs' from the CLI
08:35.58Tuskerahhh very nice
08:36.01Tuskershow codec 271
08:36.34angler-271 is a combination of several codecs
08:36.50Tuskeryeah, correct, show codec parses the mask
08:41.27Tuskerok, another question... say I am calling through a peer... but I don't hear any ring tone through that peer while it is connecting... is there any way to make a ring tone happen within asterisk until it connects ?
08:43.31angler-r option on dial
08:45.48Tuskerangler-: even after it has done the call establish?
08:46.32Tuskerie, ring while "silence detected"
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08:52.49Tuskerbtw, is it worth to purchase the g729 codec ?
08:55.24Zeeekhow many channels?
08:56.04Tusker1 to 2 channels I think should be enough
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08:57.00Zeeekfor $20 take the plunge !
08:57.36Tuskeris it legal to "try" it before buying?
08:57.47Zeeeka lot of people think it's great, my hearing is so bad I don't think it's real important but we have 4 channels (= two conversations)
08:57.58Tusker:)
08:58.20Zeeeknot as far as I know
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08:58.39Tuskerok
08:59.10ZeeekI've watched the bandwidth, it definitely saves it and it definitely sounds good
08:59.21Zeeekbut not as good as ulaw IMO
08:59.50Tuskerah ok
09:00.11coppicethe main reason to use G.729 is "the other end requires it"
09:00.27Zeeekgood point
09:00.51Tuskerbut, if the other end has ulaw, and bandwidth isn't a huge issue, then that should be fine ?
09:01.22coppiceif bandwidth is not an issue, then ulaw/Alaw is a better choice.
09:01.30Zeeekas I said :)
09:02.08Tuskerok, cool then
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09:17.44Assidisnt ulaw lossy ?
09:17.54*** join/#asterisk ToTo (n=ToTo@81.174.33.2)
09:18.48x86ulaw is lossless
09:18.54x86same with alaw
09:19.06x86that's how it is possible to do fax with them ;)
09:19.39coppiceulaw and alaw are lossy, but only a little bit
09:20.20x86no?
09:20.56Assidwell .. gotta do something.. people are complaining of having audio loss for parts of a conversation
09:22.05*** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)
09:22.37x86Assid: could be from latency
09:22.54x86Assid: also could be that they can not handle a 64kbps stream of audio
09:23.04x86Assid: try switching to GSM
09:23.11x86(if the clients support it)
09:23.16x86else, try g726
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09:23.44Assidwell.. what i did is gsm from/to provider (voicepulse) and ulaw for internal communication
09:23.56Assidbut i wonder what happens in a scenario where the calls end up in meetme
09:25.50x86Assid: "internal" == LAN?
09:27.05coppiceAssid: do they get broken audio on the LAN, or just on external calls?
09:27.49*** join/#asterisk voipaster (i=25x8supp@203.192.191.36)
09:27.50Assidwell.. people from the outside.. get broken audio..
09:27.54Assidor their audio breaks
09:27.59Assidlan people got no issues
09:28.09AssidHOWEVER.. there is ample bandwith still left
09:28.38Assidon another note.. is it possible to link to meetme locations?
09:28.41coppiceyou probably need QoS. no amount of bandwidth is adequate all the time without a little management
09:28.48Assidlike meetme on box1 and meetme on box2 ?
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09:33.33Assidbrb.. need food
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09:36.33x86Assid: it sounds like coppice has no clue as to what he is talking about (first G.711u/a being lossy, now that you can magically QoS inbound traffic)
09:36.52x86Assid: what your problem most likely is, is latency between you and your VoIP provider
09:37.03x86Assid: also, could be having jitter issues
09:37.05coppicex86: are you really this thick, or just trying to be annoying?
09:37.23x86thank god for ignore :)
09:38.27x86Assid: check latency and see if it can be improved (does the provider have a closer switch, is there an issue going on with your WAN uplink, etc)
09:39.15Tuskerx86: if there are jitter issues, what can be done about it ?
09:39.28x86Tusker: you can increase your buffers usually
09:39.45Tuskerhow would I go about doing that? :)
09:39.52x86Tusker: which means you'll increase your delay a bit, but the audio wont be as choppy
09:40.05*** join/#asterisk RoyK (n=roy@213.160.242.134)
09:40.20x86Tusker: depends on how you are connecting to your provider (TDM, IAX, SIP, whatever)
09:40.32Tuskerx86: SIP
09:40.35x86Tusker: if you're using a zaptel card, you can change some stuff in zapata.conf
09:41.27Tuskerno zaptel card
09:41.45x86http://www.voip-info.org/tiki-index.php?page=Asterisk+new+jitterbuffer
09:41.55coppiceTusker: the good jitter buffering in * is currently only for IAX. onyl recent test versions have good jitter handling for SIP
09:42.27Tuskeroh ok, so best to use IAX where possible ?
09:42.32coppices/in */in releases of */
09:43.21coppiceor wait for 1.4 :-)
09:43.42Tuskerhow much regex does jbot know ?
09:44.21Tuskers/\s(\w+)/_\s_\1/g
09:45.02coppicei think it just does simoke substitutions when it sees the s//
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09:45.28Tuskerok, cool though
09:47.28Zeeekjbot knows how to run asterisk through NAT :)
09:47.51coppiceand make coffee and doughnuts
09:47.53Zeeekjbot receives faxes in any codec
09:48.13Tuskerwow, sounds decent
09:48.20Zeeekjbot servers beer to all, but knows to serve lite to those with a weight problem
09:48.44Zeeeks/servers/serves/
09:48.45*** join/#asterisk buzzyd (n=buzzyd@82-45-247-173.cable.ubr01.enfi.blueyonder.co.uk)
09:49.06coppicedoes it distinguish those who are fat, but don't have a problem with that?
09:49.14Zeeekabsolutely!
09:50.01buzzydI have a snom 320 and whenever I try to get it to register with my asterisk 1.0.10 box I get stale nonce errors and it won't authenticate any ideas why?
09:50.11mitchelocis jbot an addon for asterisk? can someone link me...?
09:51.08buzzydCan I supply information that will help troubleshoot or is their a better place for me to ask this question?
09:51.42Zeeekput your nonces in the fridge when not in use, they're STALE!
09:52.32RoyKZeeek: fridge is full...
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09:52.43buzzydZeeek, my fridge is very full
09:52.50Zeeekremove the useless stale credentials
09:53.02RoyKlitterbox is full as well
09:53.07Zeeekfor bad password errors, make them GOOD
09:53.11coppiceRoyK: huh. the temperature rises above -20 and you start worrying about fresh food
09:53.14*** join/#asterisk fulgas (n=fulgas@209.8.233.208)
09:53.24buzzydZeeek, how do I remove stale credentials
09:53.34Zeeekit's actually about 12°C here now, very cold for the season
09:54.07RoyK15 degrees and sun
09:54.18buzzydis this an asterisk problem or snom?
09:54.18coppiceif it makes you feel any better we aren't all suffering that :-)
09:54.41Zeeekbuzzyd try a softphone to check that
09:55.10buzzydsoft phone works fine as does my snom 190 and the other 320 I've been testing with
09:55.29buzzydwhat is confusing me is I don't know what a nonce is
09:55.41RoyKbuzzyd: a hash
09:55.43buzzydhow/where its generated
09:55.43Zeeektry googling for "stale nonce"
09:55.47RoyKbuzzyd: a joint, perhaps
09:55.54buzzydwould love one :)
09:55.58Zeeekas in "see you anonce..."
09:56.09Makenshia nonce is an english word for insulting someone
09:56.17Makenshi"you nonce!"
09:56.26Zeeekbut seriously GIYF - try looking there
09:56.39Zeeekthe second link is "what is a stale nonce?"
09:56.52RoyKfor the nonce => for the present; temporarily
09:56.55Zeeekhttp://www.mail-archive.com/asterisk-users@lists.digium.com/msg114348.html
09:57.20buzzydlooking now but according to description that shouldn't affect registration
09:57.20RoyK"Ubuntu: Ancient African word for ''I'm sick of compiling Gentoo''" -- Jeff Waugh
09:57.45ZeeekOlle says the message is just a warning
09:57.59Zeeekglad I took the time to look
09:58.23coppicesagoma - the wise men of south africa. of course, the sangoma cards aren't designed or built in south africa :-)
09:58.32*** join/#asterisk vexorg (n=vexorg@CPE000021ded913-CM001225419164.cpe.net.cable.rogers.com)
09:58.55Zeeeksang=blood
09:59.02buzzydanyone here use snom 320's?
09:59.14Zeeeknah, too many stale nonces in them
09:59.19buzzydlol
09:59.21RoyKZeeek: sang == song :P
09:59.28coppiceits sangoma, not sanguine
09:59.34Zeeekthey're supposed to be great for the money though :)
09:59.41coppiceyou much be thinking of MS
10:00.00ZeeekMultiple Suckezrs?
10:00.16muti like my sangoma a104d, sorta
10:00.18buzzydZeeek, the fact it ain't login in would suggest the opposite :)
10:00.20mutit broke my faxing tho
10:00.42coppicethe sangoma cards are nice, but the installation needs a serious overhaul
10:01.01Zeeekbuzzyd could be 2 different problems
10:01.02*** join/#asterisk Vyeperman (n=Vye@ip68-6-130-59.sd.sd.cox.net)
10:01.52buzzydZeeek, please elaborate
10:01.52Zeeeklooked at the wiki? Some phones have install intructions for asterisk there
10:01.58*** join/#asterisk littlejohn (n=little@host221-76.pool876.interbusiness.it)
10:02.15ZeeekI mean maybe the nonce is a warning having naught to do with not logging in
10:04.09znoGhas anyone used the clipcomm ATAs?
10:04.57coppicehttp://www.voip-info.org/wiki/view/Clipcomm+ATA+and+Gateways
10:05.56buzzydZeeek, If it was you and snom 190, xlite and the other 320 works but this one doesn't would you think its the phone or config error?
10:07.17Zeeekyes, if you compared it it sounds that way. Next step, find a config somewhere
10:07.33Zeeekand look at sip debug
10:07.52*** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net)
10:09.05buzzydZeeek, Ok thanks for pointers off to do some more testing
10:09.42znoGcoppice: thanks, mainly interested in finding out if they have configurable dial plans in them
10:11.02*** join/#asterisk tparcina_ (n=tparcina@wr-lama.iskon.hr)
10:11.09Tuskeris it possible to do the following: incoming call into asterisk, ring extension 1 for 60 seconds, 30 seconds into that 60 seconds, try another extension, but keep extension 1 ringing ?
10:12.13znoGcoppice: good info on the products on the page you sent me, but i'm looking for peoples' experiences with them
10:16.03*** join/#asterisk acrg (n=aragon@decoder.geek.sh)
10:17.24acrgI'm looking to get a digium pri card for my asterisk setup - anyone know what is the current state of freebsd support for the cards?
10:19.38coppiceacrg: last I heard it was kinda functional, but there don't seem to be enough users to really get it shaken out.
10:20.44*** join/#asterisk edo1 (n=Miranda@pool-62-106.ptcomm.ru)
10:21.01acrgthanks
10:21.16edo1hi
10:21.52edo1exist ISDN COLP (Connected Line Identification Presentation) support in asterisk?
10:22.40*** join/#asterisk Ecio (n=eciostar@194.105.59.42)
10:23.01acrgedo CLI?  yes
10:24.28*** join/#asterisk pbx1 (n=pbx1@58.69.229.213)
10:25.13edo1no, not cli
10:26.32edo1example - i dial to asterisk box from cellular phone and see "redirected to XXX" message. XXX is bad number
10:29.14*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197)
10:30.13*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
10:31.54edo1for outgoing calls i can set callerid, what for incoming?
10:32.49*** join/#asterisk xorol (n=root@uu212-190-229-201.unknown.uunet.be)
10:33.02qdkedo1: you wanna
10:33.04qdkargh
10:33.18qdkedo1: you wanna SET callerid on incoming calls?
10:37.02tzafriredo1, you refer to bristuff's isdn specifically?
10:37.12*** join/#asterisk rstrit (n=rstrit@204.238.218.130)
10:37.46*** join/#asterisk eset (n=eset@ip545186e3.direct-adsl.nl)
10:38.40esetanyone have a suggestion why i get a error "modules/res_odbc.so: undefined symbol: ast_load" when running asterisk?
10:39.48*** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr)
10:40.05Eciohi all, im experienced bad (well, let's say not-so-good) quality with asterisk, can someone give me some hint?
10:40.12puzzledhi
10:40.21*** join/#asterisk xorol (n=dannyz@uu212-190-229-201.unknown.uunet.be)
10:40.55dtwilsonEcio: you mean poor audio quality on phones?
10:41.07dtwilsonEcio: are you using sp handsets?
10:41.08Ecioim tryin with a SIP client (x-lite) and with a cisco phone (SIP trunk between call manager and asterisk pbx) trying conference (meetme dynamic) and also echo test
10:41.10dtwilsonsip*
10:41.25Splatanyone know how well the cisco 7940 ip phones work with asterisk?
10:41.37Ecioi.e. if i try to call with x-lite (directly connected to asterisk with a sip account)
10:41.43Ecioand call echo test (or the conference)
10:41.54*** join/#asterisk zotz (n=zotz@24.231.36.9)
10:42.19Eciothe voice is not so good... but i dont think it's due to the file recording.. cause the "point" in the audio where i can see some artifact/noise change from call to call
10:42.35drraySplat - I use a 7960 cisco
10:42.40znoGthe Sipura/Linksys ATAs still seem, by far, the more feature packed ATA
10:42.41*** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)
10:42.45dtwilsonEcio: this may sound strange - but do you have power cables running parallel to the relevant network cables in close proximity? i.e. tied together?
10:44.01Eciodtwilson: i could check it... but i've tried with 2 asterisk, one running on a vmware image (on another site... passing two firewalls :D) and the other one under the desk on a real pc and on the same subnet
10:44.28Ecioi've also downloaded the audio files from astlinux.com (that afaik are supposed to be better quality) but it doesnt change
10:44.39Splatdrray: that doesn't help.. the place I have to do some work has 7940's on every desk..
10:44.49Ecioinitially i thought it was maybe a problem of timers on linux and vmware... but i have problems also on the physical machine too
10:45.03Ecioconsider that i've just installed everything, i have 0 users so no congestion, no cpu load etc..
10:45.10*** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)
10:45.49drraySplat - 7940's and 60's are the same cept the number of lines
10:45.54Ecioi've found also italian voice on tomato.it but i have this "rumors" on those too
10:46.25Eciocalling from a client to another seems to be better... but i was specially interested in the conferencing features (that together with sip trunking wit call manager could allow me to host mixed conferences with external users...)
10:47.26coppiceEcio: a common cause for this is your disks not using DMA. Unless they use DMA they block the telephony audio for sigificant periods, and you get gaps in the audio
10:47.26Assidis it possible to connect 2 meetme() of 2 different asterisk boxes together
10:47.37*** join/#asterisk lorinc (n=ang@caracas-3905.adsl.interware.hu)
10:48.53dtwilsoncoppice: I'm about to spec a box for our first real client - what kind of disks would you reccommend as best? scsi 15k rpm or sata?
10:49.04Eciocoppice : uhm... let me check it
10:49.19coppiceAssid: have you tried? it should kinda work, but I suspect the way mixing occurs might give some quality issues
10:49.21Eciodo u remember the hdparm syntax for checkin it'
10:50.03Eciousing hdparm -d /dev/hda i got using_dma    =  1 (on)
10:50.22coppicedtwilcon: SATA drives work just fine as long as they are used with DMA and with IRQs re-enabled suring processing (I forget exactly what they call thta feature)
10:50.27MrChimpymy AMI monitoring app works great, but I have no way of reading a channel variable from AMI AFAIK so I can't tell which one of our apps the caller has been routed to
10:51.01EcioUDMA modes: udma0 udma1 udma2 udma3 udma4 *udma5 <- * = active mode
10:51.09*** join/#asterisk Hadaka (i=naked@naked.iki.fi)
10:51.13Ecioso prolly it's not dma :/
10:51.25MrChimpyi go through all the channels by number from the zap channels output, then do zap channel status to each to find out if they're connected and to get the CLID
10:51.35Splatdrray: ok, so they work fine then? any tricks to them or anything?
10:51.56HadakaHello, I've got a question about Asterisk - can I use an IAX2 soft phone to talk to asterisk which would proxy it onwards as SIP to my provider?
10:53.40HadakaI know the reverse is possible (connecting with SIP to asterisk and having it use IAX outbound), but I don't know about this need?
10:55.10drraySplat - they work great, once setup for sip, you need a TFTP server for them and a smartnet contract
10:55.35Splatdrray: smartnet contract?
10:56.31Eciohave to go to dinner... later
10:57.01drraySplat - for getting firmware for the phones
10:57.37*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
10:57.43Splatdrray: ok, I wonder if they already have that.. they have about 18 or so of the phones in the office.. heh
10:57.54*** join/#asterisk supjigatr (n=syslod@152.53.17.26)
10:57.55supjigatrHi.
10:58.13drraySplat _ I'd assume so
10:58.17dpryoHadaka: That shouldn't be a problem.
10:58.21drraySplat - or maybe just one contract :)
10:58.26supjigatrAnyone know how to recover IAXy boxes after reinstall of a asterisk server without bringing them back in.
10:58.26Hadakadpryo: Great!
11:02.37Splatdrray: maybe I should just try to work out how to reprogram the cisco call manager.. (if that's actually what they have running the phones currently..) heh
11:02.58drraySplat - it depends on what they want
11:03.46edo1tzafrir: i try without bristuff, plain asterisk
11:04.05Splatneed to route calls to mobiles through a device that will let them go through the mobile network.. some calls will want to go through VoIP.. and others through the normal phone network.. heh
11:05.14MrChimpydammit, there's an AMI GetVar as it is.
11:05.21*** join/#asterisk rkr245 (n=ravi@81.21.33.35)
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11:16.31satlan32hi
11:16.36satlan32question not regarding asterisk
11:16.50satlan32tcpdump -i eth0 -s 1500
11:16.59satlan32how do i set it up to save to a file?
11:17.23*** join/#asterisk RoyKa (n=roy@213.160.242.91)
11:21.30dtwilsonsatlan32: just append '> filename.txt'
11:21.45satlan32thanks
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11:25.50muppetmasterHello
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11:36.21*** join/#asterisk Henk (n=Henk@s5593c2e9.adsl.wanadoo.nl)
11:37.03motuwhich version of asterisk do i have to get to use the hanguponpolarityswitch option in zapata.conf?
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11:37.08*** mode/#asterisk [+o denon] by ChanServ
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11:39.04HenkI have a server in a datacenter that has asterisk installed and it has budgetphone.nl registered. If i call the number I get a 'extension not found' error which is understandable because there is none, and i probably dont need one.  I'd like to get this asterisk to dial a number for me, play a wav file to the user that picks up the phone on the other side and hang up. How do I do that?
11:44.27*** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)
11:59.19*** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca)
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12:01.49tparcinacall file, - why this one doesn't call? - http://pastebin.ca/58619
12:02.15motuI get Ignoring hanguponpolarityswitch, and Ignoring signalling, why?
12:02.24tparcinai got this message on cli - http://pastebin.ca/58620
12:07.06*** join/#asterisk oej (n=oej@ip-207-145-80-8.nyc.megapath.net)
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12:09.39[TK]D-Fendertparcina : Looks lie the PSTN didn't like your number.... starting with a 0 is legit where you are?
12:11.32*** join/#asterisk fugitivo (n=ajf@190.48.166.195)
12:11.38fugitivohi
12:11.38aBd0ulaXHello, i need to know some information about asterisk for my job, someone know where i can find a good tutorial (its better in frensh)... thx
12:11.48fugitivo~docs
12:11.55jboti heard docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
12:12.24aBd0ulaXthx man ;)
12:13.13fugitivonp
12:13.27ZeeekaBd0ulaX there are several French language sites about asterisk
12:13.44Zeeekfrench-asterisk.net or something like it. Google to find them
12:13.53AhrimanesZeeek: :)
12:13.59ZeeekBeeer
12:14.23aBd0ulaXcool ;)
12:14.30Zeeekn'est-ce pas?
12:14.47aBd0ulaXhaha t français ??
12:14.51Zeeekdirectcentrex and wengo use asterisk
12:15.07Zeeekalso Acropolis Télécom
12:15.15fugitivoi'm having serious problems with a te205p, my zttest is really bad
12:15.17ZeeekAxialys
12:15.18fugitivoi tried everything
12:15.33Zeeekfugitivo cheat and you'll pass :)
12:15.50fugitivoand the motherboard is not in the incompatibility list
12:16.02*** join/#asterisk lorinc (n=ang@caracas-4553.adsl.interware.hu)
12:16.05Ahrimanesfugitivo: minor issues
12:16.32fugitivoBest: 100.000000 -- Worst: -79.895020 -- Average: 98.767997
12:16.39Eciomore on my quality test/problems i have this strange behaviour: im testing the conference, i've joined one (dinamically created) conference and the music is intermitting: when i stop talkin the music vanishes while when i emit sounds, i can here the music too... any idea?
12:16.40fugitivothat's not a minor issue
12:17.01ZeeekEcio are you using a cell phone?
12:17.07Eciono: x-lite
12:17.11fugitivoand digium support doesn't reply emails
12:17.16Zeeekmake sure xmit silence is off
12:17.30Zeeekfugitivo call digium
12:17.34Zeeekin a few hours
12:17.53fugitivoi sent 3 emails yesterday
12:18.02Zeeekwhat country are you in?
12:18.11iDunnois it nearly home time yet?
12:18.14fugitivoArgentina
12:18.29Eciozeek xmit silence is off
12:18.30ZeeekI was afraid of that fugitivo, ar is blocked on many mail servers
12:18.37fugitivo???
12:18.47Zeeektry writing from a yahoo web mail or something
12:18.59fugitivothey did send the automatic reply
12:19.02Zeeekfugitivo you didn't know AR is one of the capitals of spam?
12:19.15Zeeekok if you got that the mail came thru
12:19.58fugitivoZeeek: i don't think so, in AR we don't have enough bandwidth ;)\
12:20.45ZeeekI guarantee you a lot of spam comes from there. Not as much as China, but a lot
12:21.21ZeeekBrazil too, while we're putting down south america ;)
12:21.25Eciozeek: im tryin also from my cisco phone (via sip trunk phone ->callmanager -> asterisk conference)
12:21.35Ecioand music disappears after some seconds....
12:22.06ZeeekEcio all I know is that behavior is common on cell phones, I gues they use silence suppression for obvious reasons
12:22.28ZeeekX-Lite should work so something is indeed wrong
12:24.58sudhir492SOS - Anyone using spandsp and app_rxfax here?
12:25.06Eciozeek: the strange fact is that the music works initially, then after some seconds i can hear it only when i make sounds on my mic..
12:26.00Zeeeksudhir492 yes. rx works great sometimes, other times it won't receive a fax here
12:26.20mitchelocisn't asterfax the replacement?
12:26.38sudhir492Zeeek: Can you tell which version of spandsp and rxfax are you using?
12:26.43dtwilsonEcio: have you tried two lines from x-lite into the conference room simulataneously?
12:27.17*** part/#asterisk bartlebee (n=largo@202.5.145.13)
12:27.18Zeeeksudhir492 asterisk 1.2 and whatever the latest spandsp was at the time
12:27.26Zeeeklet me login and see
12:27.30Eciodtwilson: if i call the same room they will talk without problems (even if i call the same conference with a couple of cisco phones via callmanager-siptrunk and xlite clients...)
12:27.32sudhir492ok
12:27.46Ecioim just tryin to understand if all this small audio quality problems are fixable...
12:28.11*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
12:28.12sudhir492I tried spandsp-0.0.2pre25 and rxfax from the same directory
12:28.18*** part/#asterisk oej (n=oej@ip-207-145-80-8.nyc.megapath.net)
12:28.34sudhir492but get immediate hangup on the Zap channel
12:28.46Zeeeklooks like 2pre21 I have
12:29.01sudhir492what about rxfax
12:29.19Eciodtwilson: now i was connected with 2 clients (1 phone via trunk and 1 xlite), i disconnected the phone, now on the xlite i can here the music but with the previously mentioned problem
12:30.29Eciothat's really strange... but the worst problem is the one i wrote some hours ago, that recorded messages dont play very well...
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12:33.10ZeeekI am not easily finding rxfax version
12:33.10Zeeek(or source)
12:33.11dtwilsonEcio: You recorded messages *from* a handset or xlite etc? My guess for those the problem might simply be poor quality microphones
12:33.31ZeeekEcio have you checked for IRQ confilcts?
12:33.31Eciodtwilson: sorry i didnt explain myself, i meant the announcement
12:33.32Eciolike "this is echo test " or "this is a conference etc.."
12:33.33Zeeekif you mean sound is jumping around and uneven it's often IRQ
12:33.35Ecio"insert the conf. number and pound key etc.."
12:33.35dtwilsonahhh the room entry announcements? -gotcha now
12:33.38dtwilsonI was confused :)
12:33.47Eciodtw: sorry, bad english :)
12:34.42dtwilsonno probs Ecio :)
12:34.49sudhir492Zeeek: I see spandsp-0.0.2pre21.tar.gz
12:35.11Zeeekthat's what I have. app_rxfax.c is dated Nov 2005
12:35.25sudhir492I am going to try the same version. Will you please email me app_rxfax.c and app_txfax.c
12:35.37Zeeekare they no longer on the site?
12:35.40*** join/#asterisk RoyKa (n=roy@213.160.242.91)
12:35.41sudhir492my email address is sudhir492@gmail.com
12:35.47dtwilsonunfortunately I'm stumped now though Ecio - I was earlier presuming you mewant you had echo problems with voice audio
12:35.57dtwilsonmeant*
12:36.11Eciozeeek: the jumping audio is heard also in the cisco phone (connected to call manager)
12:36.20ZeeekIRQ?
12:36.32Zeeeksudhir492 I don't have or use txfax
12:36.51sudhir492thats fine. rxfax is the important one for me
12:38.22Zeeekisn't it on the site? It isn't easy for me to mail it right now
12:39.30Zeeek<PROTECTED>
12:40.21Eciozeeek i've tried to switch back to original english sound, but the problem is persistent so i dont think that the italian voices are broken...
12:40.29esetanyone had experience installing asterisk from source on a debian 2.6 kernel?
12:40.38ZeeekHAVE YOU CHECKED IRQ ?
12:40.46esethaving trouble with the ztdummy driver
12:41.41dtwilsonEcio: check for IRQ problems on the server as per Zeeek's suggestion
12:41.54Eciozeeek: what can i do to check them? (btw i have problems on a vmware machine and on a physical machine too.. so two completely different hardwares)
12:43.11[TK]D-FenderZeeek : They're there...
12:43.15ZeeekIRQ is physical
12:43.33Zeeekso cat /proc/interrupts and see
12:43.50Zeeekyes, mine are too [TK]D-Fender
12:44.03ZeeekANd they be lookin' real good!
12:44.11Eciozeeek: btw i dont have any digium card...
12:44.39Zeeekok, that never occurred to me :)
12:45.19Zeeekassuming there are no IRQ problems with ethernet (and assuming there is an ethernet interface) that would not be IRQ
12:45.34Eciohttp://pastebin.ca/58630
12:45.57[Airwolf]Is it possible to force Asterisk to always use a single ip address in the sip headers when registering ?
12:46.21Eciozeeek: those are the two /proc/interrupts... i dont think there are conflicts
12:46.43[Airwolf]Because I have a server with two interfaces and somehow Asterisk registers himself with the ip adres from the other interface.
12:46.55dtwilsonairwolf: you tried bindaddress=ip.ad.dr.ess ?
12:46.55[Airwolf]What disables any communication ofcourse
12:47.13Zeeekdtwilson way too simple and logical
12:47.17dtwilsonin your sip registration
12:47.39[Airwolf]dtwilson, I ding it to 0.0.0.0 right now. Because I want to use both interfaces for SIP communication.
12:48.13*** join/#asterisk tdonahue-laptop (n=tdonahue@64.201.13.172)
12:49.01Eciozeeek: if u want i can try to create one test account and give it to you so u can tell me if it's normal the quality  i got or not...
12:49.26ZeeekEcio I'm sorry, no time for that
12:49.38Eciook np
12:51.10*** part/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com)
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12:57.18dtwilsonairwolf: another stab in the dark might be to try using fromdomain=whatever.domain.com in your sip registration context and set that specific domain in etc/hosts to the ip of choice
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13:00.53protocoldougfor three way calling, do you use the MeetMe() command in your dial plan?
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13:06.47asteriskstercisco 7960 with SIP ver 8.2 getting hanged once in a day during call transfer i am using asterisk ver 1.2.5/zaptel-1.2.5
13:07.15drrayasteriskster - why did you upgrade to 8.2?
13:07.33drrayI'm just curious
13:07.57asterisksteractually i have changed it firmware SCCP to SIP
13:08.45*** join/#asterisk cybergypsy (n=mark@APoitiers-156-1-51-239.w86-217.abo.wanadoo.fr)
13:09.27asterisksterdo u know any firmware which is stable and reliable on heavy load 150-200 calls per day
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13:11.32dpryoAre there any easy ways to inject audio on open channels? (Programaticly..)
13:14.30asteriskstercan anyone have stable cisco 7960 SIP firmware works in heavy load to share with me
13:15.18docelmoasteriskster if your asking for someone to give you the firmware why not ask but considering we all pay for access to the firmware I doubt it.
13:15.38asterisksterok fine
13:15.42asteriskstersorry
13:15.48asterisksterif u mind
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13:19.11nokyhi
13:20.29JackEStormbaah, this sucks, stupid ass queue
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13:21.32Assidthere has to be a way to link 2 meetme conference rooms from 2 different boxes together
13:21.42[TK]D-FenderJackEStorm : As this seems to be the first thing you've said in the past hour at least perhaps you could tell us what the problem is?
13:21.44*** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.235.155.Dial1.SanJose1.Level3.net)
13:21.54noky[TK]D-Fender: hi
13:21.57nokyhttp://pastebin.com/734977
13:22.12[TK]D-FenderAssid : Issue a call-file for it
13:22.12nokyi have realtime for extensions and sip_buddies implemented in my asterisk
13:22.38nokybut i don't know why appears some querys to sql repeated.....
13:22.51JackEStorm[TK]D-Fender: first thing I said in days at that :) ...the app_queue rings members if they are in use, and I got someone bitching about that.
13:22.56Assidhrmm.. that could be a good way to do it actually
13:23.33[TK]D-FenderJackEStorm : set them to no-call-waiting, and 1 call limit on the phone level.
13:23.54nokyis common ?
13:24.18[TK]D-Fendernoky : Bridging MeetMe's?  Doubtful
13:24.56nokyi have meetme... but i aren't using now...
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13:25.02Assidnoky: its a very good way to cut down bandwith usage tho
13:25.40JackEStorm[TK]D-Fender: got that like that now to shut them up for now, but thats not going to work, because I need to allow unlimit outbound, allow people to direct dial and extension and have it ring even if they are on the phone, but have the queue only send calls to that member if they are not on the phone.
13:26.20[TK]D-FenderThe thing is I'm not sure how DTMF will travel between them in case audible in-band gets processed, and I see no way short of having the admin kick the bridge down for it to close when you're finished.
13:26.26Assidimagine this.. 10 people from location A .. 20 people from location B  want to be in conference. what do we do? 10 goes to meetme in A, 20 go to meetme in B.. then we link them together.. so its only 1 call between the 2 locations
13:27.19[TK]D-FenderJackEStorm : Plan "b" : use AgentCallbacklogin and have the script check if they're on the phone before actually sending the call through.  The worse you'll get is a jump in the distribution cycle.
13:28.20[TK]D-FenderAssid : Yes, a very profitable idea.  you can also have 1 SIP phone call both MeeteMe's and then conference them 3-way and force a re-invite.
13:28.20fugitivo[TK]D-Fender: i couldn't solve the problem :(
13:28.40fugitivo[TK]D-Fender: i'm returning the card
13:28.48*** join/#asterisk ToyMan (n=stuq@74-32-76-147.dsl1.mdl.ny.frontiernet.net)
13:28.51Assidyes, but if that person needs his phone.. for whateevr reason. we are using up the lines
13:29.20Assidalso phone level versus pbx level, im guessing you might get more quality output with the pbx's calling each other
13:30.07Assidjust gotta figure out if we can make it such when they call a particular extension..  it initiates a call between the 2 boxes
13:30.49[TK]D-Fenderfugitivo : You know where to go now....
13:30.59Assidor.. if you have multiple offices, what we can do is have a "initiating" call.. which links our box to the remote's meetme
13:31.19[TK]D-FenderAssid : No... force a re-invite and that'll biridge the 2 conferences he's on and release his phone.
13:31.24*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
13:31.46Assidphone gets released?
13:32.08[TK]D-FenderAssid : the 2 calls get bridged by * after that.  Its in the more recent polycom firmware.
13:32.24Assidwouldnt hhe need to use the conference facility of the phone for it to work?
13:32.27[TK]D-FenderRead your Release Notes!
13:32.49*** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)
13:33.27[TK]D-FenderAssid : Thats what I said... caller on side "A" call's "A"'s meetmet, then "B"'s meetme and conferences local to the PHONE.  Then he hangs up completely and the 2 calls (bot MeetMe's) are rebridged together
13:34.03Assiddamn.. i thought if you hand up your phone, the conference dies
13:34.05*** join/#asterisk Inkubot (n=inkubot@200.119.229.247)
13:34.08Assidhang even
13:34.34Inkubothi
13:34.41Assideven if its re-invite
13:34.43Inkuboti'm having a problem with my sip trunk
13:34.44[TK]D-FenderAssid : You aren't listening -> WITH THE MORE RECENT FIRMWARE <-
13:34.56docelmohiya!
13:34.56Inkubotwhen i'm recieving sip calls trough a SIP trunk, i get this
13:35.10[TK]D-FenderAssid : READ THE RELEASE NOTES!  They are just for starting fires you know!
13:35.12InkubotMay 24 10:35:59 NOTICE[30544] chan_sip.c: Failed to authenticate user "5629582667" <sip:5629582667@200.74.178.XXX>;tag=as1ad15b75
13:35.13Inkubot5629582667 is the number that calls
13:35.20Inkubotwhy my Asterisk try to authenticate this number as a user ?
13:35.30Assidso how would you kill the "conference" the phone starts?
13:36.15[TK]D-FenderAssid : you don't.  The remain permanently bridged.  A meetMe admin may be be able to kick the link if its the only one left.
13:36.15Assidalso wouldnt this only work for SIP based OR if the same phone registers with BOTH the boxes
13:36.43[TK]D-FenderAssid : wouldn't matter which...
13:37.25Assidcurrently i have sip phones -> * <-> IAX <-> * <-> SIP phones
13:37.44Assidphone on A .. would need to dial through local * box to jump to the other box
13:38.10Eciodtwilson & zeeek: it looks like the problem was/is related to the virtual machine, now i've created another trunk from the call manager and calling from my cisco phone to the conference running on asterisk@physical_machine the audio seems to be a lot better
13:38.14*** join/#asterisk BadPacket (n=root@unaffiliated/badpacket)
13:38.14Assidi think i need to learn more on reinvite
13:38.26vgstershould sip debug peer XXX debug all sip traffic?
13:38.31vgstercos it is on my system
13:39.09esetis there anyway to run meeting rooms without a zaptel module?
13:39.25vgsterah ignore me figured it
13:39.32MrChimpyztdummy?
13:39.55esetthats a zaptel module isnt it?
13:39.56[TK]D-FenderAssid : Basically it doesn't matter HOW the phone makes its way to Server B.  Once there, the local * has both calls flowing through it and thats where the rebridging takes place.  Its not a literal "re-invite" in a peer-to-peer sense.
13:40.38eseti am having probs getting ztdummy to work on debian
13:41.13Assidi see
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13:42.46Inkubotejejje i solve the problem
13:42.51Inkubotinsecure=very
13:43.03Inkubot: )
13:45.03Assidjust curious tho.. if a sip phone connects to another sip phone.. and both share the same codecs.. would it use the preference of the caller/callee?
13:45.29Inkubotif a good thing to have insecure=very ?
13:45.33Inkubotif/is
13:46.14*** join/#asterisk C4T3l (n=rcall01@216.54.143.2)
13:46.21Ariel_morning everyone
13:46.57[TK]D-FenderAssid : "preference"?
13:47.24docelmoARIEL!
13:47.28docelmo:)
13:47.43docelmoGod I am so bored..
13:48.26noky??
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13:48.32*** mode/#asterisk [+o anthm] by ChanServ
13:48.32nokyhttp://pastebin.com/734977
13:48.33noky:(
13:48.33Ariel_docelm0, sorry to hear it.
13:48.49*** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br)
13:49.43Assid[TK]D-Fender: doesnt canreinvite get affected if both parties behind nat?
13:49.46esetwonder if any deboan asterisk guru can help me, i have  a 2.6.8-2 kernel and zaptel is compiling and storing in /lib/modules/2.6.8  not  /lib/modules/2.6.8-2 ...if i simply move the module it says incorrect format
13:49.47[TK]D-Fenderdocelm0 : One of my old English teachers used to say "boring is between your ears"
13:49.52*** join/#asterisk cjk (n=cjk@80.92.64.103)
13:49.59[TK]D-FenderAssid : You aren't listening again....
13:50.04*** part/#asterisk Inkubot (n=inkubot@200.119.229.247)
13:50.06[TK]D-Fender[09:39] <[TK]D-Fender> Assid : Basically it doesn't matter HOW the phone makes its way to Server B.  Once there, the local * has both calls flowing through it and thats where the rebridging takes place.  Its not a literal "re-invite" in a peer-to-peer sense.
13:50.11Ariel_noky, sorry your using real time. I don't know or use it.
13:50.15[TK]D-FenderAssid : Read the last sentence.
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13:50.27cjkhi, how good is the asterisk spool for .call files. what happes when i put 10k files in it
13:50.59Ariel_10k wow that is allot. Depends on your box. But that just might over load it.
13:51.04tzafrircjk, I guess they'll be processed in turn.
13:51.15[TK]D-Fendercjk : Use the "mv" method to shift them in and I guess it should work.... just a question of concurrency which I couldn't say....
13:51.19docelmoUsually there is something going on but its just dead today
13:51.33[TK]D-Fenderdocelm0 : Why do you think I'm here :)
13:51.35Assidoh.. i thought if a call was done via sip uri straight to the 2nd box.. would be different
13:51.36tzafrirChances are asterisk will not handle easily 10000 concurrent calls.
13:51.48Assidmy bad
13:52.24*** join/#asterisk Lino` (n=Lino@i577BCDCA.versanet.de)
13:52.36cjkok, anyway to set a maximum limit so that asterisk processes them when the other calls are finished
13:53.26*** join/#asterisk bkw_ (n=brian@adsl-70-142-54-60.dsl.tul2ok.sbcglobal.net)
13:55.07[TK]D-FenderAssid : Don't use a straight URI
13:55.10*** join/#asterisk mko-025 (n=korpim@p5498B28C.dip0.t-ipconnect.de)
13:55.28*** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net)
13:55.30[TK]D-FenderAssid : make sure both calls pass through THAT PHONE'S local server.
13:57.01Assidbut if i disable the ulaw codec from my sip.conf.. and only allow 729 .. then it uses 729.. this is even though the phones preference and sip.conf preference is 729
13:58.12filemeep?
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14:00.56jaikequick question guys. anyone experiencing crashes using mixmonitor?
14:02.00wunderkinjaike, what version are you using
14:02.09jaike1.2.7.1
14:02.24*** part/#asterisk kmilitzer (n=km@office-gw.westend.com)
14:02.24jaikeits not very often..maybe once every 1000 calls
14:03.19wunderkintheres nothing in bugs, get a bt and post it
14:03.47Eciois there any call manager expert here? im tryin to strip a digit for calls between CM and Asterisk but i have no success
14:03.59wunderkinunless you try the 1.2 branch, i know there have been some fixes but dont know if anything past 1.2.7.1
14:04.22jaikeour servers processes around 5000 calls a day, that amounts to around 5 crashes..but going back to old monitor, we dont have problems..except for the huge spike in CPU utilization
14:05.09wunderkinupdate first, make sure it is producing core files, if still happens then report
14:05.41jaikehmmm..ok will do that..thanks
14:07.14MrChimpyhey guys
14:07.54MrChimpyi really need to ask a quick question to a digium person, anyone around? we've just got 3 TE411Ps so I reckon we've paid to have a quick question answered :)
14:09.03AssidMrChimpy: ask the question if anyone can help.. they just will
14:09.50MrChimpywell, what does the ID switch do on the card? do I need differing settings per card if I'm putting two in the same box?
14:12.19sevardriddle me this
14:12.21brettnemhey anyone in here using the SIP jitterbuffer?
14:12.42sevardyou have a four line phone yet a line now and then goes UNREACHABLE
14:12.56brettnemMrChimpy: If you put more than one of those cards into the same box you will need to have unique ID.
14:13.21brettnemMrChimpy: BTW, buying equipment from Digium has nothing to do with anyone's willingness to answer a question. ;)
14:13.45brettnemsevard: what kind of phone is this?
14:14.02sevardAastra 480i CT
14:14.23sevardit happens on Sipura 2002 ATAs too, one line will go unreachable for 20 or some odd minutes
14:15.06blitzragebrettnem: no -- but hopefully you will, test, and provide feedback so it can get into 1.4.x -- but I doubt there is enough time now for it to get in
14:15.25fileblitzrage: we are really trying to get it in
14:15.34fileit's just... Russell and I tried yesterday, and the trunk version is toast
14:15.46fileseems like everyone has been testing the 1.2 version
14:15.49brettnemblitzrage: I have a lot of production servers and I'd like to test it out, but I'd like to get some idea of how usable it is right now
14:15.52MrChimpybrettnem: thanks :)
14:16.13brettnemsevard: that is weird. maybe a nat issue? Do they all use the same port?
14:16.31sevarddifferent ports behind a switch NAT'd once
14:16.32brettnemsevard: btw, I was just trying to get my first aastra 480i CT up yesterday coincidentially
14:16.44sevardbrettnem: they're great phones
14:16.48brettnemsevard: might be a port translation issue (broken router)
14:16.55sevarddo you want an example cfg file for your tftp server?
14:17.15brettnemsevard: I haven't been avle to get it to dial yet... it recieves calls just fine.. but when I dial, it just gives me dialtone back
14:17.28brettnem<dialtone> <digit> <silence><digit><dialtone>
14:17.29sevardthat... weird
14:17.37brettnemyeah
14:17.39sevardcheck your firmware version, i'll check against mine
14:17.56brettnemok, it's a customer's phone, I'll ask when they get in.
14:18.12esethi, I wonder if any Debian asterisk guru can help me, i have  a 2.6.8-2 kernel and zaptel is compiling and storing in /lib/modules/2.6.8  not  /lib/modules/2.6.8-2 ...if i simply move the module it says incorrect format
14:18.19sevardApplication: Version 1.3.0.1080 SIP  Boot ROM: Version: 1.1.0.4
14:18.35brettnemsevard: I'll check
14:18.42sevardbrettnem: alright, if you need config files to provision it hit me a line when they come in
14:18.42blitzragefile: yah - I read something about that yesterday... thats really good bad :(
14:18.51sevardI spent a day figuring out all the neat little bells on this guy
14:19.08*** join/#asterisk gcarrillog (n=gcarrill@201.152.19.192)
14:19.11brettnemsevard: looks neat, but I'm not sure what advantage it has over a nice cordless phone and a sipura
14:19.30sevardbrettnem: 9 lines, four cordless RF phones
14:19.37brettnemsevard: that is nice
14:19.39sevardbrettnem: big buttons, nice display
14:19.52sevardbrettnem: it's WAY easier to xfer and conf and intercom on this phone than regular phones
14:19.58brettnemsevard: I haven't actually seen it yet.. my customer is like 600 miles away ;)
14:20.05sevardbrettnem: services, directory listing, awesome quality intercom
14:20.15brettnemcool
14:20.20sevardbrettnem: :) i love that phone, my favorite part though is the weight in the handset
14:20.36sevardbrettnem: just the perfect amount
14:20.36Assidhey [TK]D-Fender, any update on the 1.6.6 when polycom plans to release it for us 'regular' folks?
14:21.01brimstoneeset, did you get your 2.6.8-2 problem fixed yet?
14:21.28esetbrimstone : no
14:21.45brettnembbiab
14:21.56*** join/#asterisk sb_mx (n=sb_mx@200.94.154.226)
14:22.29brimstoneeset, set EXTRAVERSION to = "-2" in /lib/modules/`uname -r`/build/Makefile
14:22.36brimstonethen recompile zaptel
14:22.44esetah
14:23.43filebrimstone knows too much, hit him!
14:24.08brimstone<zoidberg> look! i'm helping!
14:24.11esetbrimstone : thanks, i will see how this goes and let you know :)
14:24.23brimstoneokey dokey eset!
14:28.00*** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197)
14:31.53esetits official, brimstone does know too much :)
14:32.06brimstoneoh noes!
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14:32.48eset(thanks)
14:33.08brimstoneyou're welcome eset
14:33.25brimstonedoes anyone happen to have the original IVR recording from the song Lame by Tool?
14:35.19*** join/#asterisk Henk (n=Henk@s5593c2e9.adsl.wanadoo.nl)
14:36.59HenkI'm trying to use phpagi (2.14) to script my way into asterisk, But I got stuck at the example part, if i run an instance of the class i get an error about port 5038.. what should be running on that port ?
14:37.41jake1932manager should be running on that port
14:37.54jake1932is manager enabled in manager.conf?
14:38.59jake1932btw - agi is different from manager
14:39.31*** part/#asterisk buzzyd (n=buzzyd@82-45-247-173.cable.ubr01.enfi.blueyonder.co.uk)
14:39.51Henkjake1932, Ah ok I'll take a look. I did a 'grep -Hir 5038 *' to find where the port was mentioned but i guess i made a typo ... did not see this one before
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14:43.42*** join/#asterisk jaike (i=jaike@210.5.119.146)
14:43.47Henkjake1932, hmm... I'm just getting to know all this today... it's a big toolset. What i'm trying to set up is an interface to the web for a phonenumber verification system (you enter your number on our website, we call you with a pin number, you enter the pin number on our website, we know you are not a person hiding his identity)
14:44.11*** join/#asterisk Skarmeth (n=Skarmeth@200164212156.user.veloxzone.com.br)
14:44.19mutsweeeeeeet
14:44.19muthttp://www.evilchili.com/mediaview/1614/Apache_Disco_Video
14:44.26*** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net)
14:44.33Henkthe php-agi interface seems perfect
14:44.41*** join/#asterisk darkskiez (n=darkskie@194.247.78.146)
14:45.33*** join/#asterisk Delta239 (n=none@201.226.130.55)
14:45.45Supaplexif only it gave you cash back
14:45.54Skarmethhi all
14:46.25blitzrageheh... you can't find girls that skinny in the USA anymore
14:47.56jake1932Henk: you can drop a .call file when the request comes from the web
14:48.21Eciois there a tool/command in CLI that shows what dialplan is applied when u try to call a number?
14:49.15Eciosomething like "what happens when i dial 4666 from here?"
14:49.20blitzrageEcio: the context that is applied when you dial depends on how you authenticate, which then processes the call in the context assigned to that user/peer (in sip.conf / iax.conf, etc...)
14:49.39blitzrageEcio: no -- what happens depends on what you have in the dialplan
14:49.42SkarmethWhen I instruct the Dial command for example to do Dial(${EXTEN:1}), it removes the first digit of dialed extension (MSD), if I need to do something like remove all digits of ${EXTEN} except the first 2 digits, add w041 and get ${EXTEN} again and remove only the 2 first digits and use the resulting string, it should look like it or I need another app (spaces just to turn more readable)? Dial ( $ { 2 : EXTEN } w 0 4 1 $ { EXTEN :
14:49.42Skarmeth2 } )
14:49.43jake1932Ecio: set verbose 4
14:49.46Eciouhm
14:50.07jake1932Ecio: or higher
14:50.24blitzrageSkarmeth: ${EXTEN:length:offset} is the format
14:50.59blitzrageSkarmeth: to keep the first two digits, you do: ${EXTEN::2}
14:51.07Skarmethblitzrage, like ${EXTEN:2:1} ?
14:51.13blitzrageSkarmeth: sure
14:51.13Skarmethok
14:51.35SpaceBassthis disco video is AWESOME!
14:51.49blitzrageEXTEN=ABCDEFG  -> ${EXTEN:2:1} -> BC
14:52.05blitzrageto go from the other end... use negative numbers
14:52.06Skarmethblitzrage, when it is done, ${EXTEN} value continues unchanged right? it just a copy of it...
14:52.09mut:)
14:52.16blitzrageSkarmeth: yes, ${EXTEN} is unchanged
14:52.16Henkjake1932, and this .call file can contain an interactive session (welcome message, type a # to confirm, spit out the code, spit out the code again, record in the system that it went OK and that the user has received the code)
14:52.19Skarmethblitzrage, thanks
14:52.54blitzragethat disco video sucked
14:52.55jake1932Henk: nope - you can use the .call file to point to a point in your dialplan to process the call
14:53.10blitzragejake1932: you use the Local/ channel for that
14:53.36jake1932blitzrage: he wants to call it from a web app
14:53.40blitzrageahhh
14:54.22blitzragehttp://www.youtube.com/watch?v=dMH0bHeiRNg
14:57.08Henkjake1932, ah google just pointed me to the docs. Seems to be OK for starters, but i definately want to use the php api, the possibilities for an interactive session between the webinterface and the real person on the phone are just too good not to try.
14:57.42Henkcombined with ajax the user will be completely baffeled
14:58.08*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
14:58.44jake1932Henk: there's plenty of cool stuff to do - just seems for your app, dropping a simple .call file would work just fine
15:00.31Henkjake1932, yep for a start that will be OK. But i'd like to do more in the future. We are a VPS provider and I would like to do stuff like having nagios (a monitoring server) make a call to a customer telling hem his server is down, and giving him some options like reboot etc
15:01.03jake1932Henk: we already do that (the same way)
15:01.29Henknice... what company ?
15:01.39jake1932Henk: the .call file just start the call - you'll throw the meat in the dialplan
15:01.46Delta239how can i access the home directory on asterisk?
15:01.55jake1932Henk: a big cable company :)
15:02.07esetokeokde, now i get a saddening "Invalid module format" for ztdummy when i modprobe it on 2.6.8
15:02.15Delta239the only thing i have here is 4 files with no folders
15:03.13Henkjake1932, yep i saw the .call achitecture, its just a way of calling out and once succesfull bind the call to a regular extension.
15:03.37*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
15:04.11jake1932Henk: correct.  we use the manager API also, but it serves a different purpose.
15:04.23*** part/#asterisk jaike (i=jaike@210.5.119.146)
15:04.24jake1932totally different application
15:05.00Henkdo you do this with php? java? ..
15:05.05jake1932.net
15:05.51jake1932we made a custom app to interface directly through port 5038
15:06.28*** join/#asterisk tiwyant (n=twyant@pix.wyantcomputerservices.com)
15:06.48tiwyantMornin'
15:07.44*** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net)
15:08.00asterboygood morning...good morning...good morning the whole day through...
15:08.07tiwyantthat too
15:08.28tiwyantAnyone around know if there's a limit on how many mailboxes I can include in a group voicemail?
15:08.42asterboyI took some VIagra, CIaliz, and something else that gave me 500% cum volume...good morning!
15:08.47Henkjake1932, I see. For php I found a class that abstracts a few things for me and lets me hook the commandset of asterisk right into my existing adminsoftware. So once I get it to work I guess i should be able to do some nice things
15:09.05jake1932Henk: correct
15:09.59jake1932asterboy: i just got that same e-mail this morning
15:10.03jake1932asterboy: and yesterday
15:10.11*** join/#asterisk blebleble (i=godie@caesar.godie.net)
15:10.19asterboylol...I get them every 5 min
15:10.54mercestesWhy would I want 500% cum volume?  Wait.....nevermind....don't tell me.  I dont' wanna know.
15:11.08Henkjake1932, if I have a closed dialplan for incomming connections and I have manager listen on 127.0.0.1 and i do checks on the stuff php sends to asterisk. Are there any security issues i must address to get the default asterisk safe?
15:11.11asterboydrown her in your pleasure!
15:11.12MrChimpydunno. girls get thirsty
15:11.51*** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com)
15:11.54jake1932Henk: are you exposing asterisk on the net?
15:12.08mercestesmeh.....I didn't wanna know, I said.
15:12.11MrChimpyweird. i've installed 2nd TE411P, it works and I can dial to it, but zap status shows Alarms on the working port as NOP
15:12.32jake1932Henk: IOW - i have a mythtv setup with a bunch of ports open that's completly safe
15:12.40bleblebleanyone ever run into flash operator panel not showing all the extensions, it fills up the page yet i have a lot more its now showing
15:13.45jake1932Henk: it's not accessible to the public internet though
15:13.45asterboyAnyone here experienced with paging systems?
15:13.46Henkjake1932, for incomming calls yes. I want to just redirect calls to one of our 24-hour support staff members' cellphones after office hours.
15:13.46esetanyone know why the svn of zaptel gives a "module format invalid" when you try to modprobe the zaptel modules?
15:13.46tiwyantwhat kind of paging?
15:13.47asterboysimple, handset to paging horn.
15:13.48nokyi have asterisk with realtime's extensions... and i don't know why the log /var/log/asterisk/full logs 3 querys to my mysql database each time
15:13.52nokyany idea?????
15:14.17*** join/#asterisk prog (n=prog@vdsoft.kh-net.cz)
15:14.21tiwyantasterboy:  Find yourself a nice fxo/fxs capable paging amp and do it that way.  Works like a charm.
15:14.25proghello to all
15:14.47tiwyantblebleble:  Have you edited the op_buttons.cfg and changed the sizes?
15:14.53asterboyya, I have the LUPCM Bogen
15:15.12asterboyDo you use an FXO or FXS to hookup the paging system?
15:15.12jake1932Henk: who are you protecting the system from, callers?
15:15.21asterboyI'm thinking FXO
15:15.28progif someone says: "our SIP provider doesn`t require registering" - what does it mean ? How can I solve this in sip.conf ? ( register=> IP_address ) ?
15:15.29bleblebletiwyant: i have not, i see the defaults which ones should i be changing?
15:15.31tiwyantHmm
15:15.46jake1932Henk:  website visitors?
15:15.48tiwyantasterboy:  I think FXO is how I set the last one up.
15:16.06asterboyya, otherwise the paging system is going to need to figure out what to do when called.
15:16.10tiwyantblebleble:  lemme check, real quick
15:16.32bleblebletiwyant: thanks there just is a lot of crazy options, rectangle etc etc
15:16.38tiwyantasterboy:  The last one I set up that way just answered the call and let me talk.
15:16.54jake1932Henk: re: phpagi, i'm not familiar with the security issues
15:16.59asterboythat was fxs then if it answered.
15:17.37asterboyor was it fxo and then when you go off hook it makes the connection.
15:18.23*** join/#asterisk azzie (n=az@azzie.net)
15:18.36*** join/#asterisk nortex (n=nortex@ama-wldhcp.696130103.amaonline.com)
15:18.42tiwyantblebleble:   under op_style.cfg you need to modify the scale and margins.  Just play around with it and every time you save changes reload op_panel and then your browser window.  It took me about 2 hours to get it right but it looks good.
15:19.08tiwyantasterboy:  I'm pretty sure I had an fxo port into it because I was using the other port for backup 911
15:19.16tiwyantbut the device was switchable
15:19.21asterboyah
15:19.26asterboyI think mine is also.
15:19.36asterboyThe other solutino is to go out the sound card
15:19.42tiwyantyuck
15:19.52tiwyantI'd rather go fxo/fxs and not screw with the sound scard
15:19.54asterboybut then, I'll need an amp and run speaker wire
15:19.57tiwyantyeah
15:20.00*** join/#asterisk slobberknocker (n=ckwall@63.149.122.94)
15:20.05asterboyya, I agree
15:20.13asterboymore to go wrong at the * side
15:21.22asterboytiwyant, what did you have for the speakers?
15:21.31tiwyantquick question:  Anyone know what permissions /var/spool/asterisk should have?  I'm getting errors saying the message attribute files aren't there (msg0000.txt) when they are but they aren't accessible
15:21.46tiwyantasterboy:  In ceiling whatever jobbies they had with the Merlin system I replaced
15:21.58asterboyMine either wants an amp or 70V speakers, which I'm guessing need to be powered
15:22.12tiwyantyes
15:22.31asterboySure like to get a unit that has its own amp.
15:22.40tiwyantI agree, that's the way to go
15:22.44Henkjake1932, i was thinking about the possibility for any person (calling or visitor) to do stuff like having my asterisk forward his call to expensive numbers etc
15:23.26jake1932Henk:write a tight dialplan and any callers or callees won't be able to do anything
15:24.05asterboyThis one looks good: http://cgi.ebay.com/Bogen-Telephone-Paging-Amp-TPU-15A_W0QQitemZ9730190914QQcategoryZ51279QQssPageNameZWDVWQQrdZ1QQcmdZViewItem
15:24.22*** join/#asterisk ToTo (n=ToTo@81.174.33.2)
15:24.25slobberknockerok, i am back with another stupid question i cannot find the answer to... I have multiple contexts in my extensions.conf. [TK]Defender helped me get that working. Well now i am having trouble understanding how to make it so that I can use multiple context within my zapata.conf, I have tried just adding multiple lines context=x but that did not work, and I am not finding anything helpful on the web.
15:24.26Henkjake1932, great. Well off to home now. workday is over. thanx for all your input. I'm pretty sure I can get this to work tomorrow
15:24.44jake1932Henk: good luck with it
15:24.44blitzrageslobberknocker: it doesn't work like that
15:24.51asterboyah forget paging...I'm going to start my own CLEC: http://www.voip-info.org/wiki/view/How+to+start+a+Clec
15:24.51slobberknockerok
15:25.11tiwyantCLEC = lots of paperwork and crap
15:25.18tiwyantI'm just going to start a phone sex line
15:25.19jake1932[TK]Defender lol
15:25.25tiwyantusing voip over the internet to india
15:25.34bleblebletiwyant: 755
15:25.46tiwyantthanks blebleble
15:25.54asterboyok, forget clec...I hagte paperwork
15:26.00slobberknockerwell the trouble i am having is that I have did's in each context. but only the dids in the context listed in my zapata.conf are working.
15:26.13asterboyback to 500% cum volume sales
15:26.19slobberknockeri get the error, exten 6406 not found in context progrexion.
15:26.24slobberknockerit is under evolution
15:26.32slobberknockerso how do i specify?
15:27.05Delta239anybody here knows about astguiclient?
15:27.24tiwyantIn your extensions.conf do you have an exten => entry for 6406 in the progrexion context?
15:27.24tiwyantCause the error you posted says you don't
15:27.26jake1932slobberknocker: [progrexion] doesn't have an extension 6406
15:27.46slobberknockerright... it is under a different context.
15:27.48jake1932you need to modify your extensions.conf file
15:27.56slobberknockerbut i have to specify a context in zapata, right?
15:27.59[TK]D-Fenderjake1932 : Whats so funny?
15:28.04slobberknockerwell i will have multiple contexts.
15:28.13slobberknockerhow do i make the zapa.conf use all of them?
15:28.49jake1932nick changing
15:28.56jake1932it is already
15:29.12[TK]D-Fenderslobberknocker : You don't.  Typically you send INCOMING calls to either the same context (treat all incoming calls identically), or seperate them by line (give each line its OWN contex)
15:29.35jake1932slobberknocker: you just need to include exten => 6406,1,Something in your progrexion context
15:30.11bleblebletiwyant: hay for the scale and margin is that on the icon, led, or arrow ?
15:30.16slobberknockerok, so you are suggesting that if I have did's assigned to people, that I should put them all into something to the effect of an [incoming calls] context?
15:30.54jake1932slobberknocker: yes
15:31.06slobberknockerok, great. Thanks for the help
15:31.14jake1932thank defender
15:31.42*** join/#asterisk wunderkin (i=kev@69.26.192.234)
15:31.54*** join/#asterisk xy_goat (n=hotjokb@pdpc/supporter/student/xy-goat)
15:32.11jake1932[TK]D-Fender: where does your nick come from?
15:33.15*** join/#asterisk _Paulo_ (n=Paulo@c90621fa.virtua.com.br)
15:33.56[TK]D-Fenderjake1932 : Originated from myplaying Tribes 1 CTF.  I tended to "watch the fort", hence the nick.
15:33.56MrChimpyfrikkin weird. even zttool says NOP for the single span I'm actually using yet it still works
15:34.21jake1932[TK]D-Fender: ok
15:34.34[TK]D-Fenderslobberknocker : Ok, maybe we should start a little further back.  What kind of technologies are you using with your server?
15:35.06[TK]D-Fenderjake1932 : [TK] was my old Action:Half-Life clan.
15:35.10slobberknockerare you meaning am i using a T1, etc?
15:35.17[TK]D-Fenderslobberknocker : yes
15:35.29slobberknockeri have a TE410P with 2 pris
15:35.42[TK]D-Fenderslobberknocker : Describ every kind of device taking calls into your system, VoIP pproviders, T1/E1/Analog, everything
15:36.13slobberknockerall providers are T1 and all devices are polycom 301 and 501s
15:36.54[TK]D-Fenderslobberknocker : Ok, you should be sending ALL channels to the same context and then telling each DID where to go.  You may reserve DID's for "direct to employee" purposes, some for differnt classes of IVR's (customer service, general, sales, ect), and so on, but only *1* context referenced in Zapata.
15:37.07[TK]D-Fenderslobberknocker : Excellent phone choices.
15:38.22slobberknockerok, so stop me if i am heading in the wrong direction then... what I am about to do is create an inbound and and outbound context. on the outbound context i am going to add my dial plan [outbound]
15:38.22slobberknockerexten => _1XXXNXXXXXX,1,Dial(Zap/g2/${EXTEN:1})
15:38.22slobberknockerexten => _1800NXXXXXX,1,Dial(Zap/g2/${EXTEN:1})
15:38.22slobberknockerexten => _NXXXXXX,1,Dial(Zap/g2/801${EXTEN:0})
15:38.34slobberknockerand on the inbound i will do all of my dids and ivrs.
15:38.58slobberknockerthen on those dont i have to do something to the effect of include => {other contexts}
15:39.11[TK]D-Fenderslobberknocker : yes you 100% want to seperate contexts for in/out... you don't want people calling IN to have access to call OUT do you?
15:39.16slobberknockerthe main goal is for tenanting. i need to have different companies on one system
15:39.28[TK]D-Fenderslobberknocker : Since I now know you're on PRI your dialplan needs a MASSIVE overhaul.
15:39.35[TK]D-Fenderslobberknocker : PM
15:39.39slobberknockerok
15:42.57*** join/#asterisk s0lid (i=s0lid@gr-153-200.eglobalreach.net)
15:44.29*** join/#asterisk kaz0358 (n=kaz@kazg5.telecom.ksu.edu)
15:44.38Ecioguys i have a working trunk between asterisk (1.2.4, a@h???) and a call manager and im tryin to setup another trunk between another asterisk (1.2.7, a@h2.8) and the same CM.
15:44.52EcioI can call from the CM to the asterisk but not viceversa, i got a "Dial failed due to CHANUNAVAIL" in the debug
15:45.11Eciothe strange thing is that i see this in the debug of the not working one:
15:45.11_Paulo_~a@h
15:45.12EcioDial("SIP/60666-170e", "/4666|120|W
15:45.20kaz0358quick question, if you do not have reverse dns going.. will that cause problems for some sip servers when they try completing a call to your asterisk box? i want to say that i had problems with that earlier, but i'm not entirely for certain. anyone have experience with that?
15:45.38[TK]D-FenderEcio : that dial statement is VERY wrong...
15:45.45Ecioyeèp
15:45.48Eciothe working one is
15:45.51EcioDial("SIP/70002-2658", "SIP/CCM1/4666")
15:46.05nokyany expert of realtime extensions of asterisk ????
15:46.05[TK]D-FenderEcio : And I doubt anyone here is going to want to hear about your systems running A@H.  Please read the channel topic.
15:46.08Ecioi cant understand what's happening.. i've just copied the macro.. :)
15:46.40xy_goathi all - how can i stop jitter from happening on my asterisk install?
15:46.51Eciod-fender: actually im fed up of this a@h... most of the times when u do something in the web interface everything's messed...
15:47.15kaz0358ecio, the configuration files aren't that hard to manage..
15:47.15Eciou try to edit something in the console, and the web int doesnt work anymore
15:48.05_Paulo_Ecio, thats pretty much why people here dont like to support *@h
15:48.34Ecioi see
15:50.02Eciobut on asterisk.org there are only sources, not binaries, isnt it?
15:50.16kaz0358would someone like to help me out with a quick test phone call? i'd like to confirm that the reserve dns thing is indeed causing problems
15:50.33[TK]D-FenderEcio : A@H = Craptastic cookie-cutter config generator thats nigh-impossible to debug
15:50.36*** join/#asterisk zotz (n=zotz@24.231.36.9)
15:50.45*** join/#asterisk Mw3 (i=mw3@national.t-error.hu)
15:50.58CunningPike[TK]D-Fender: Tell us how you really feel ;)
15:51.09Eciod-fender :)
15:52.13[TK]D-FenderShould have checked your magazine!
15:52.21blitzragebooo
15:52.21[TK]D-Fender:D
15:52.24blitzrageroll playing sucks :)
15:52.30blitzragerole* even
15:53.04_Paulo_Ecio, most linux distros have * packaged
15:53.13[TK]D-FenderActually I'm just waiting for the new forge models to arrive before I prepare to buy my next one...
15:53.24Eciouhm, so i'll take my time to install a debian and try asterisk on it
15:53.33_Paulo_I use Debian, you just have to type "apt-get instal asterisk"
15:53.33Katty[TK]D-Fender: beep!
15:53.35esethmmm
15:53.35[TK]D-FenderEcio : And for the love of God avoid those packages LIKE THE PLAGUE
15:53.43eseti would think about that twice (debian and asterisk)
15:53.44[TK]D-FenderKatty : ! ! !
15:53.45Eciod-fender which packages ? a@h ? :D
15:53.45[TK]D-Fender;)
15:53.49blitzrageugh-- don't use packages... learn how to build Asterisk -- and it's not even hard
15:53.52Katty[TK]D-Fender: you set off my hilight.
15:53.57*** join/#asterisk tamp4x (n=Lab@64.201.13.172)
15:54.09blitzrageand I hate debian, so what do I know? :)
15:54.09sevard:(
15:54.12sevardi don't get any hugs anymore
15:54.13tamp4xdoes asterisk support options messages?
15:54.15[TK]D-Fender[11:53] <Katty> [TK]D-Fender: you set off my hilight. <- ooohh the things you say...
15:54.17esetbut good luck if you need zaptel
15:54.47[TK]D-Fenderblitzrage : You hate Debian?  Very respectable distro.  Just avaoid with a very few particular pckages and its gold...
15:54.47esetdebian is giving me monstrous trouble wiht module formats for zaptel, eek
15:54.57sevardDoes anyone have any experience with DirectVNC?
15:55.32myiagyi have quite a few systems running debian and asterisk.. with 2.4 and 2.6 kernels.. including on a AMD64.. never had much trouble
15:55.56blitzrage[TK]D-Fender: in my opinion Debian is a bitch just to be a bitch -- and the <quote>hackers</unquote> seem to like it that way
15:55.57esetno trouble with building the zaptel drivers on 2.6?
15:55.58Ecioeset: i dont need zaptel, i just need sip (and maybe iax) clients, conferences and trunk with call manager :)
15:56.08_Paulo_eset, I followed the recipes for debian @voip-info.org and everything worked well.
15:56.23esetEcio : you need zaptel for the conferences even with sip
15:56.25Ecioblitz: if debian is a bitch, what's gentoo?  :D
15:56.27Katty[TK]D-Fender: pfft.
15:56.28myiagyeset hm, no.. it build just fine..
15:56.47Ecioeset: dont they use "virtual" modules?
15:57.10[TK]D-FenderKatty : ;)
15:57.10esetztdummy is a virtual module for zaptel as far as i can tell, wont modprobe with 2.6
15:57.18Eciolol
15:57.32blitzrageEcio: a whote -- I hate gentoo too
15:57.39blitzrage-t +r
15:57.44*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.38.245)
15:57.48_Paulo_eset, I dont use ztdummy, I have a digium board.
15:57.49fileI like Debian.
15:57.55blitzragegentoo is for people who have far too much time
15:58.05Ecioblitz :)
15:58.06[TK]D-FenderGentoo is for ricers!
15:58.08esetit seems there are a few peole though having the same problem
15:58.17esetlooks like it is somewhat historical
15:58.24tamp4xwhy would i get a 404 not foudn when i recv and options message
15:58.29asterboyno, lfs is for people with too much time.
15:58.39eset2.6 gives a "invalis module format" error for zaptel modules
15:58.51Kattyeset: :<
15:59.01sevardi love you
15:59.01sevardfag
15:59.04Katty[TK]D-Fender: i think i'm about to put my very first production box up at a client's.
15:59.15mercestesGentoo is cruel and unusual.
15:59.16Katty[TK]D-Fender: i'm all skeered inside :<
15:59.18myiagyeset i got that error once. but i think the problem was with gcc version
15:59.23mercestesI love you too, Sevard...
15:59.23mercesteshomo.
15:59.26esetwith gcc, really?
15:59.33asterboybrokeback *....again!
15:59.41mercestesGentoo is text based chinese water torture.
15:59.41myiagykernel was compiled with gcc 4.. and i compiled zaptel with gcc 3
15:59.45esetyou mean a diff gcc version from the one used to compile the kernel maybe
15:59.48esetah, yes
15:59.53esetok, i will look at that
15:59.56Ecioeset: so what distro do u suggest
16:00.02asterboyLFS!
16:00.03esetum....
16:00.07esetatari?
16:00.09eset;)
16:00.11asterboylol
16:00.13Ecioaster: i've not enought time...
16:00.15Ecio-t
16:00.26Eciobut maybe on UZIX on MSX
16:00.34Ecioit could handle 1 sip call in half duplex mode
16:00.37esetim going for paper cups and string
16:00.37Ecio=)
16:01.01esetmost reliable telephoney platform there ever was
16:01.25mercestesI advocate kororaa.org
16:01.28Ecioeset: uhm and being practical? :D
16:01.30mercestesI run it.
16:01.30[TK]D-FenderKatty : Doing this for a business now?  Not just intenal consumtion?
16:01.59mercestesit's gentoo based without the torture, crying and suicide attempts.
16:02.06Katty[TK]D-Fender: yes'm :>
16:02.08esetEcio : if debian works with thise gcc tip then i'll let you know
16:02.13Katty[TK]D-Fender: :> :< :> :<, etc.
16:02.53[TK]D-FenderKatty: Your personal consulting on on behalf of your company (as I'm not really sure where you work)
16:03.47Katty[TK]D-Fender: it's a leetle place.
16:03.50esetseems my kernel was compiled with the same version of gcc, so dont think tahts it
16:04.10filebbl
16:04.18Kattykbi
16:04.30*** part/#asterisk slobberknocker (n=ckwall@63.149.122.94)
16:04.55[TK]D-FenderKatty : So was that "you consulting for you" or "you doing it for your job"?
16:04.55myiagyeset i think the gcc error says something about "version magic" so yours might not be the same error i got
16:06.11esetok, hmmm
16:06.24esetit was a good idea though
16:06.44myiagyi had the invalid module thing too.. just can't remember what it was.. i'll tell you if i remember..
16:07.29myiagyeset do you have kernel-headers installed?
16:07.33esetyep
16:07.48myiagywhat version of kernel you running
16:07.55eset2.6.8-2
16:08.51*** join/#asterisk slobberknocker (n=ckwall@63.149.122.94)
16:09.16esetmyiagy : tried with module-assistant, as well as svn, and the latest tar
16:09.28esetand apt-get of course
16:09.32*** join/#asterisk vechers (n=103326C9@64.61.117.138)
16:09.37Katty[TK]D-Fender: oh.
16:09.41Katty[TK]D-Fender: it's for a company, not me.
16:09.51Katty[TK]D-Fender: though i'm really the only one here that knows anything about asterisk.
16:09.57Katty[TK]D-Fender: and, admittedly, that ain't a whole lot at all :<
16:10.59esetmaybe i just try a diff gcc anyway
16:11.07myiagyeset can you copy the whole error?
16:11.45eseti use this: modprobe ztdummy, and i get this error:
16:11.55esetWARNING: Error inserting zaptel (/lib/modules/2.6.8-2-686/zaptel/zaptel.ko): Invalid module format
16:12.08esetFATAL: Error inserting ztdummy (/lib/modules/2.6.8-2-686/zaptel/ztdummy.ko): Invalid module format
16:12.13esetFATAL: Error running install command for ztdummy
16:12.19RaYmAn-Bxeset: check dmesg
16:12.41esetzaptel: no version for "struct_module" found: kernel tainted.
16:12.47esetzaptel: version magic '2.6.8-2 SMP preempt PENTIUM4 gcc-3.3' should be '2.6.8-2-686 preempt 686 gcc-3.3'
16:12.58RaYmAn-Bxit's that last one
16:13.00*** join/#asterisk Coyotee (n=root@sipx.ica.net)
16:13.01esetyeah
16:13.03esethmmm
16:13.16mercesteskernel tainted.  I like that error.
16:13.16*** join/#asterisk wiseguy_ (n=chivilis@infospalvos.lt)
16:13.21wiseguy_hellow
16:13.25Coyoteehey all
16:13.29HymieI'm using asterisk 1.2.1... and it thinks that 10 seconds is 15 seconds... that is, when it says "nobody answered the call in 15000 ms, it's really been closer to 10.  As well, when I have someone in my queue, it thinks that 600 seconds is about 3 minutes.. any ideas?
16:13.43esetso how do i work around that?
16:13.55Hymieeset: you can use modprobe -f to force it to insert anyhow
16:14.13wiseguy_how do i write extension if i want to execute system command after call?
16:14.21eset-f doesnt work, same error
16:14.26Hymieeset: er
16:15.00RaYmAn-Bxeset: using a distribution kernel or self-compiled?
16:15.15Coyoteedoes anyone happen to know where i can find a decent installation tutorial for asterisk on fc4?
16:15.23*** join/#asterisk Holos (n=asdf@204.101.26.106)
16:15.27myiagyeset the headers you have installed
16:15.30esetdist kernel
16:15.34myiagyare not the same for the kernel you have running
16:15.46esethmmm
16:15.52myiagyi think thats what this means: zaptel: version magic '2.6.8-2 SMP preempt PENTIUM4 gcc-3.3' should be '2.6.8-2-686 preempt 686 gcc-3.3
16:15.54RaYmAn-Bxmyiagy: they shouldn't be either (supposedly)
16:16.20myiagythey shouldn't?
16:16.30HolosAnyone know if I can I use my sangoma and PRI to dial out to a modem and create a serial connection? I need to dial into a computer and my modem isn't working.
16:16.32myiagydon't you need smp headers if you have an smp kernel
16:16.46RaYmAn-Bxthat might be true
16:16.48esetthe version is (from /proc/version : Linux version 2.6.8-2-68
16:17.03wiseguy_help me, someone
16:17.06esetand the headers are kernel-headers-2.6.8-2
16:17.29myiagytry updating the headers then
16:17.32esetok
16:18.21*** join/#asterisk mut (n=animenod@65.111.222.120)
16:18.27muton a dark desert highway
16:18.29prog21:14:24.162830 IP 192.168.0.250.5060 > 113.151.64.101.5060: UDP, length: 359
16:18.29prog21:14:24.176512 IP 113.151.64.101.51782 > 192.168.0.250.5060: UDP, length: 358
16:18.30mutcool wind in my hair
16:18.40mutwarm smell of something, rising up through the air
16:19.05progwhy opposite site ( 113.xxxxx ) replies with 51782 port ?
16:19.33progthis can`t work
16:20.04mercestesmut:   when off in the distance..I saw a shimmering light.
16:20.17mutso i called up the captain, please bring me my wine
16:20.19mercestesmut:  my head grew heavy and my sight grew dim...I had to stop for the night.
16:20.32mut:P
16:20.39muteveryone together now!
16:20.45[TK]D-Fendermercestes : Oh the two of you can stop ANY TIME now...
16:20.50mutliviin it up at the hotel california!
16:20.58mutheh
16:21.07mutman i wish today would end already
16:21.12muthour and a half and i'm gone
16:21.42*** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com)
16:21.48esetwell, it cured the first error
16:21.58wiseguy_ghem
16:22.01*** part/#asterisk Holos (n=asdf@204.101.26.106)
16:22.02wiseguy_someone, help me
16:23.05esetmyiagy : it works!
16:23.07wiseguy_i have a problem writing extension. I wan't to execute System command after end of all calls in that context?
16:23.31esetif i knew who u were and i had any money you'd be in my will ;)
16:23.42myiagyeset ;)
16:23.46esetthans
16:23.54esetthanks, that saved me a real headache
16:24.57esetnow i'm gonna document it to save someone else a headache
16:25.01sevardDoes _anyone_ have either fbvnc or DirectVNC working
16:25.09mercestesHey, Wiseguy.  Guess what I found under google "asterisk extensions execute system command.
16:25.11mercestesThis link
16:25.12mercesteshttp://www.voip-info.org/wiki/index.php?page=Asterisk+-+documentation+of+application+commands
16:25.16sevardhaha
16:25.36mercestesMay the Google be with you, young padawan.
16:25.48mercestesIronically, the command you are looking for is called System.
16:26.13mercestesCheck out this link too.  http://www.catb.org/~esr/faqs/smart-questions.html
16:26.24slobberknockershouldnt it be that when i have one phone in one context, they should not be able to dial a phone from another context?
16:27.01wiseguy_mercestes: okay, show me the answer, because i haven't found there something special
16:27.10mercestes...............
16:27.20mercesteswell.....the little letters......
16:27.21*** join/#asterisk Qwell[] (i=north@unaffiliated/qwell)
16:27.25mercestesif you slur them all together, make words.
16:27.46mercestesand those words convey ideas.....and the idea is ....search the page for hte word System.
16:27.54*** part/#asterisk kaz0358 (n=kaz@kazg5.telecom.ksu.edu)
16:29.51*** join/#asterisk flujan (n=flujan@internet.nube.com.br)
16:30.07*** part/#asterisk xy_goat (n=hotjokb@pdpc/supporter/student/xy-goat)
16:30.10flujanping coppice
16:30.21wiseguy_mercestes: i not asking how to execute the command, i'm asking how to write exact extension for all ended calls
16:30.23mercestesslobberknocker:  Technically, if you had an extension in one context and not in another context, you should not be able to dial that extension from the not having context.
16:30.38wiseguy_mercestes: i mean s, 103, o what?
16:30.40flujanguys, i want to configure asterisk to work with a legacy proprietary pbx system.
16:30.57flujanFirst of all, I plugged the E1 interfaces using a crossover T1 cable.
16:30.58mercestesslobberknocker:  The phone itself is arbitrary in that statement.
16:31.11flujanthen, I compile asterisk to work with the mfc r2 signalling.
16:31.27mercesteswiseguy_:  specifically after answered calls?
16:31.49mercesteswiseguy_:  Or after all calls period regardless of handling?
16:31.54flujanThe digium card's led just flash a red light... should it be green?
16:32.13wiseguy_mercestes: no, after all clauses, i want the "fact" about ended call..
16:32.21flujananyone here already see a woriking  E1 card?
16:33.03jake1932i had one working
16:33.23jake1932you can use zttool to see if you have any alarms
16:33.35flujanjake1932, have it a green light ?
16:33.47flujanjake1932, yes, zttool give no errors.
16:34.00flujanjake1932, how do you have your zaptel.conf?
16:34.16jake1932i'm not running it anymore in e1 mode - using t1 now
16:34.50jake1932when running zttool, and you get a call, are you noticing any bit flips?
16:35.04mercesteswiseguy_:  Could do phpAGI, and have it hang until the call is done and deliver the handling code to you.
16:35.56mercesteswiseguy_:  or you could do a  Goto(s-${DIALSTATUS},1)
16:35.56mercestes<PROTECTED>
16:36.32mercesteswiseguy_:  But if I recall correctly, a Answered/Hangup() scenario will terminate at the hangup() and not continue priority jumping.
16:36.45flujanjake1932, http://pastebin.com/735315
16:36.57flujani got the above message when I try to make a call.
16:37.45JunK-Yzap show status
16:39.34mercesteswiseguy_:  I think using AGI is going to be the only real way to handle it for an answer situation, but I could be mistaken.
16:39.53flujanJunK-Y,  cat /proc/zapte/1
16:40.09flujanJunK-Y, can I type this instead?
16:40.46Hymiedoes anyone know why a caller would just sit in a queue, and never timeout?
16:40.53flujanJunK-Y, someone here already configure a E1 card?
16:40.56JunK-Yflujan: actually isnt the same output.
16:41.16JunK-Ynot me, since im in North America.
16:41.42flujanJunK-Y, http://pastebin.com/735335
16:42.35flujanJunK-Y, A E1 link has 30 channells... When I configure it, the configurations spans to a second channell.
16:42.36jake1932<PROTECTED>
16:42.38JunK-Yso ur span2 is all r ight.
16:43.06flujanjake1932, sorry, I'm only using the cat /proc/zaptel/1
16:43.32flujanJunK-Y, I have just one E1 link. the span 2 is not connect to the other computer.
16:43.47flujanTE406P
16:43.51flujanI have this card
16:44.06HmmhesaysJunK-Y
16:44.14JunK-Yhey Hmmhesays !
16:44.22flujanjake1932, how can I debug about the alarm?
16:44.30Hmmhesaysbeen awhile, how are you?
16:44.39*** join/#asterisk salviadud (n=ralfalfa@dsl-201-129-72-124.prod-infinitum.com.mx)
16:44.41JunK-Yim fine, summer is comin', ya?
16:44.44jake1932flujan: start with a loopback connector
16:44.45*** join/#asterisk BugKham (i=CKGLOB@125.24.7.45)
16:44.54Hmmhesaysabout damn time it si
16:44.55Hmmhesays*is
16:45.06Hmmhesaysyou're coming to cluecon aren't you?
16:45.13JunK-Yprobably yes.
16:45.15JunK-Yu?
16:45.16*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)
16:45.53BugKhamany news about 1.2.8?
16:45.54Hmmhesaysyeah I think so
16:45.56flujanjake1932, just to confirme
16:46.04flujanjake1932, to confirm...
16:46.08flujanjake1932, http://www.jaredsmith.net/misc/cables/
16:46.42flujanjake1932, I used a straight connector and a T1 crossover is it right?
16:46.44salviadudhas anyone here been to Defcon?
16:46.47jake1932http://kb.digium.com/entry/1/95/
16:47.34*** part/#asterisk BugKham (i=CKGLOB@125.24.7.45)
16:48.01flujanjake1932, thanks
16:48.17flujanjake1932, I just want to make sure about the crossover T1. Is that correct?
16:48.48jake1932crossover cable != loopback connector
16:49.37flujanjake1932, and to configure a E1 interface: http://pastebin.com/735358
16:50.09jake1932flujan: did you make a loopback connector according to http://kb.digium.com/entry/1/95/?
16:50.20flujanjake1932, not yet
16:50.53flujanjake1932, Must I start from a straight t1 cable and apply that changes?
16:51.00jake1932it's good to have that for troubleshooting
16:51.22jake1932flujan: you can butcher a straight t1 cable
16:51.35jake1932just make sure you have a spare :)
16:55.13Hymiedoes anyone know why a caller would just sit in a queue, and never timeout?
16:55.31Hymieit doesn't matter what time I pass to the queue() command, the caller sits forever in the queue!
16:56.01*** join/#asterisk u168138 (n=u168138@fangio.ee.port.ac.uk)
16:57.06u168138evening all, i`m trying to connect two asterisk boxs using sip, has anyone done it before?
16:57.36sevardwow, looks like they have a whole lot of speakers lined up for this cluecon
16:57.48*** join/#asterisk tsurk0 (n=tsurko@85.187.160.157)
16:57.57*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
16:58.59Delta239how do i modify a command like to edit a file
16:59.29*** join/#asterisk masonf (n=masonf@dungle.vineyard.net)
17:00.05AndyCanybody got 2 asterisk boxes connected together using SIP before?
17:01.02Hmmhesayssevard stfu and gbtw
17:01.38masonfis it possible keep the pstn ringing after a call gets to a zatpel card?
17:01.39sevardHmmhesays: lickamaballsa
17:03.05Hmmhesaysi was an 1 1/2 late for work this morning
17:03.15C4T3lgreat scott!
17:03.20jake1932slacker
17:03.29Hmmhesays<shrug> they don't pay me enough
17:04.00mercestesHow does that make you unique??
17:04.59sevardupi
17:05.08sevardyou were one and a half late for work?
17:05.36*** part/#asterisk masonf (n=masonf@dungle.vineyard.net)
17:06.22sevardDelta239: uhh, what
17:06.57Delta239what is the command to modify a file
17:07.07sevarduse whatever editor you want
17:07.13sevardvi, vim, emacs, pico, nano
17:07.17Delta239and how do i exit from the editor
17:07.21Delta239ok i used vi
17:07.24sevardwhat fucking editor are you using
17:07.25mercestesI suggest vi.
17:07.28sevard:q!
17:07.32mercestesIt's the easiest.
17:07.34sevardI don't suggest vi
17:07.38sevardpico is easier.
17:07.43Delta239how do i exit vi
17:07.45sevardwayyyyyyyyyyyy easier
17:07.47mercestesVi is easier.
17:07.54jake1932Delta239: look up
17:07.55sevardDelta239: press escape and type q!
17:07.58[TK]D-FenderTastes great!  Less filling!
17:08.16sevardmercestes: dude, the learning curve for pico/nano is like 2.2 seconds, for vi you litterally have to look up commands to learn it
17:08.24Delta239and does it automatically saves
17:08.31mercestessevard:  Not if you know what the commands mean in it's mother language of klingon.
17:08.36sevardDelta239: use wq to save and exit
17:08.50mercestessevard:  Which only makes sense since VI is clearly the klingon word for "Edit."  HELLO!?!
17:08.50sevardmercestes: f0r teh ubar n3rd
17:09.05Delta239thanks
17:11.59*** join/#asterisk vijatit (n=vijay@61.11.90.90)
17:14.33vijatitCan someone tell me how to vary the sampling interval for iax calls?. I need to do this to reduce bandwith requirement of the call.
17:15.28*** join/#asterisk masonf (n=masonf@dungle.vineyard.net)
17:16.32*** join/#asterisk aze (n=aze@ACayenne-101-1-12-192.w81-248.abo.wanadoo.fr)
17:16.35sevardDoes anyone know how to limit the amount of channels each sip / iax client can use?
17:20.30*** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982)
17:20.40_Paulo_vijatit, what codec are you using?
17:21.17[TK]D-FenderPolycom IP 430 appears to have displaces the IP 501 as my first-run business phone choice :)
17:21.18vijatitgsm
17:21.27*** join/#asterisk lorinc (n=ang@caracas-4553.adsl.interware.hu)
17:21.42[TK]D-FenderLikely a category killer....
17:22.11_Paulo_vijatit, change to a codec with better compression is not an option?
17:22.37vijatitI have used the "packetization=" directive for sip clients and have made calls at low bw
17:23.18vijatitactually, the codec requires only 16kbps, but overheads require 40kbps
17:23.42_Paulo_if bandwidth is of concern, better pick a codec that target this problem.
17:24.09[TK]D-Fendervijatit : You are TRUNKING that IXA link I hope....
17:24.27Delta239sevard: when i finnish putting the lines there on vi editor... i pressed ESC and when i try to type w and then q and nothing
17:24.31Delta239it wont exit vi
17:24.36*** join/#asterisk jayk- (i=jayk@lasziv.reprehensible.net)
17:24.38vijatiteven with g729, the required bw is 8kbps for data and 32kbps for ip/udp/rtp overheads, as there are frequent packets.
17:24.50jayk-is there a way to put the asterisk configuration into a pgsql database and have it read from that?
17:24.56mercesteslol
17:25.14mercestesDelta239:  Vi does not coddle the weak!  Here is what you do..
17:25.21*** join/#asterisk ToTo (n=ToTo@host135-167.pool872.interbusiness.it)
17:25.37mercestesDelta239:  Tap esc twice.  press :   (That's ; while holding the shift key)
17:25.46mercestesDelta239:  Enter wq           and press enter.
17:25.46sevardDelta239: press escape and type :wq
17:25.57vijatitbut, if we reduce the packetization frequency, the data still is 8kbps, but the ip/udp/rtp overheads come down to 8kbps
17:26.00[TK]D-Fendervijatit : Are you TUNKING your IAX calls or not?
17:26.02Delta239ahhhhhhhhhhhhhhhhh
17:26.04Delta239:D
17:26.09Delta239xD
17:26.14mercestesDelta239:  See, isn't klingon easy?
17:26.15vijatitthat is the sole idea of packetization directive in sip
17:26.21Delta239yeah hehe
17:26.25Delta239thanks
17:27.00vijatitD-Fender, actually, this is for calls from clients. These calls come as one call per dest and hence cant be trunked
17:27.10mercestesDelta239:  I would say your welcome but klingon has no such expression.  So, english wil have to suffice.
17:27.23sevardDelta239: seriously, this isn't the place for vi help.  Any retard who spends 3 seconds on google could have gotten that answer.  Either start to google or pick up nano.  If you don't attempt to find the answer yourself first you will get NOWHERE here.
17:28.02_Paulo_Delta239, type :help and go throug the tutorial.
17:28.05mercestessevard:  ......it took me longer than 3 seconds........*cries*  That's just hurtful.
17:28.31jake1932"Any retard who spends 3 seconds"
17:28.43jake1932lo
17:28.46jake1932l
17:28.48_Paulo_Delta239, vi has a very steep learning curve, but it pays after years... :-)
17:28.55[TK]D-Fendervijatit : Oh well.... with SIP you can set the frame size from 20 ms to whatever you want assuming the client can match and that will seriously reduce your overhead.  SIP vs IAX is a moot point if you aren't trunking.
17:29.12mercestesHere ya go.  http://www.thinkgeek.com/homeoffice/mugs/7bbe/
17:29.26[TK]D-Fendervijatit : So moving to 40ms packets chops your overhead in half
17:29.50vijatitD-Fender : ya. thats the point i was making
17:30.03vijatitnow, is there a way to do the same with iax?
17:30.54*** join/#asterisk diclophis (n=diclophi@65.203.37.58)
17:30.58diclophishello all
17:31.12vijatitD-Fender : With sip, at this stage, there is a patch available and after applying htat, its just a single configuration addition in sip.conf
17:31.14diclophislets say i have 2 PRI 'log terms' plugged into two seperate asterisk boxes
17:31.20diclophisis it possible to dial one from the other?
17:31.20[TK]D-Fendervijatit : no clue... should be I would think...
17:32.04*** join/#asterisk gunk (n=cch123@64.89.118.139)
17:32.35_Paulo_diclophis, you will have to create a context in both boxes.
17:32.38Delta239whatever dude i thought i could ask you guys but thanks anyway... next time i won't ask this types of questions
17:33.20diclophis_Paulo_ yea i have both boxes setup to accept calls for their respective terminated numbers
17:33.23mercestes:(  Now see what you did, Sevard...you hurt his feelings.
17:33.28*** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg)
17:33.30diclophishowever when i try dialing from one machine to the other, it wont connect
17:33.55_Paulo_how are you dialing?
17:34.05diclophisand i get a bunch of "Don't know what to do with control frame 15" messages
17:34.16diclophisjust a standard dial originate through a manager api
17:34.30vijatitThe packetization section in page http://www.voip-info.org/wiki/view/Asterisk+codecs says that asterisk supports "packetization=20ms only" for RTP based channels. there is no indication for iax.
17:34.50diclophisand i get a "PROGRESS with cause code 31 received"
17:34.57diclophisthats on the dialing machine
17:34.58_Paulo_vijatit, what is on the client side?
17:35.05diclophisthe answering machine never registers any call
17:35.17vijatitan iax softphone
17:35.49_Paulo_vijatit, your softphone supports speeks?
17:36.23vijatitboth ends are under our control, both interms of code and configs. Currently, it doesnt have speex, but i could add it
17:36.33vijatitwill adding speex help?
17:38.27_Paulo_vijatit, I think it pays just try it out.
17:39.26vijatiti'm ready to use speex, if it inherently has a longer/cusomizable packetization time. Paulo, do you know of any details?
17:40.08_Paulo_see http://www.speex.org/comparison.html
17:41.22flujanjake1932, the loopback interface works
17:41.40flujanjake1932, I have two green leds
17:42.15diclophiswould it make any difference that the 2 machines are in the same building?
17:42.19jake1932flujan: great!
17:42.24flujanjake1932, http://pastebin.com/735449
17:42.26*** join/#asterisk Peaceful (n=Peaceful@70.98.162.62)
17:42.40jake1932flujan: much better
17:42.41flujanjake1932, why appears the span 3
17:42.51flujanjake1932, since I just have two spans configured
17:42.55Peacefulcan you set values from the telnet interface on a Cisco 7960 phone?
17:42.55Nuggettelnet is eeeeeeevil!
17:43.07jake1932flujan: looks like you have 3 coofigured
17:43.58*** join/#asterisk caio1982_ (i=caio1982@CAcert-br/caio1982)
17:45.08*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
17:45.29flujanjake1932, http://pastebin.com/735457
17:45.37flujanjake1932, just two...
17:45.53flujanjake1932, this is a strange behavior, isn't it?
17:46.11jake1932indeed
17:46.42diclophishere is an interesting part too, when i try to dial the other machine i get this message from the telco "Your call cannot be completed as dialed"
17:46.49jake1932flujan: someone else may have more insight on why it is happening
17:47.07sevardOH MY GOD IT FRIGGEN WORKS
17:47.21*** part/#asterisk slobberknocker (n=ckwall@63.149.122.94)
17:47.22flujanguys, i configure just two e1 links in my asterisk box... in the zap show status appear three configured
17:47.33flujansomeone can help me with that?
17:49.03flujanjake1932, I tried to place a call I receive this: http://pastebin.com/735462
17:50.21*** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154)
17:50.23gbodemantvhi
17:50.41gbodemantvis anyone using multiple servers and one central VM location
17:51.24diclophisas in hosting one voicemail extension on a central server that is connected to other servers through iAX?
17:52.29gbodemantvI have 3 servers, each with its own voicemail
17:52.56gbodemantvproblem is that the voicemail on one does not see to want to talk to the others
17:52.57diclophisoh, but the voicemail dir is hosted on NFS?
17:53.05gbodemantvit was
17:53.16gbodemantvtried that
17:53.18diclophisyea, the app_voicemail with asterisk sucks
17:53.31gbodemantvbut it started serious lag and problems
17:53.37gbodemantvhad to rollback to local voicemail
17:53.42diclophisare you using realtime mysql config extension?
17:53.47gbodemantvyes
17:54.02diclophiswhat is the problem?
17:54.47gbodemantvwith nfs, it seemed to work, but when people would dial in, the prompts were very very slow, and then it takes forever to get the beep
17:54.53gbodemantvthen just hangs up
17:55.04gbodemantvthe NFS box is across fiber to another office though
17:55.11gbodemantvI think that is the issue
17:55.34_Paulo_gbodemantv, use something like rsync
17:55.50jake1932<PROTECTED>
17:56.39*** join/#asterisk bzbw (n=wlwzhang@ip67-153-142-109.z142-153-67.customer.algx.net)
17:56.39_Paulo_gbodemantv, syncronous methods will hurt latency
17:56.41PeacefulAnyone know the cisco syntax to set a value from the telnet interface on a 7960? (assuming you can do such a thing)
17:57.00flujanjake1932, yes, i will first try to understand why I configure 2 and appears 3 configured E1 link
17:57.04mercestesPeaceful:  Try links.
17:57.35mercestesPeaceful:   It's a command line HTTP client similar to telnet, with mouse compatibility, that will let you edit that 7960 via an interface similar to telnet.
17:58.01gbodemantvis it better just to have all servers sent to my main server for voicemail
17:58.03bzbwhi, anyone know if i can use this:  exten => _6XXX,hint,SIP/${EXTEN}  ?  show hint gives me empty result
17:58.16Flautohi people, how does senddtmf work?
17:58.21diclophisgbodemantv, if i were using the default asterisk voicemail i would have it hosted off one machine
17:58.27Flautolike if i dial a phone number with ivr
17:58.29Strom_Cbzbw: you need a priority
17:58.36diclophismainly because of how asterisk stores the voicemail files
17:58.46Flautowait for a few seconds and senddemf with the extension number?
17:58.49diclophisyou could be running into NFS locks or something
17:58.50gbodemantvand just have the other 2 servers refer to it in extensions.conf?
17:58.59bzbwStrom_C: So if I give it a priority, it will work?
17:59.00diclophisyea, through a IAX connection
17:59.09Strom_Cbzbw: supposedly
17:59.11diclophisif i am not mistaken with IAX you can make it seem transperent
17:59.23Strom_Cyou need to get your extensions language syntax correct :)
17:59.36mercestesSTrom_C:  hints do not have a priority.
17:59.43mercestesstrom_C:  That I am aware of.
17:59.48Strom_C*shrug*
17:59.52diclophislike somehow merge the dialplans from the boxes
18:00.13mercestesbzbw:  I would try entering in a few "hints" without the wildcards and see if that works for you first.
18:00.14bzbwStrom_C: thx.
18:00.24Strom_CI'm just half-assedly glancing at the channel while on the phone with the telephone company
18:00.42*** join/#asterisk Assid (n=assid@203.115.83.214)
18:00.57mercestesStrom_C:  Weren't you that guy that was mean to me earlier?
18:01.09mercestesyea..I remember you.
18:02.30flujanjake1932, how can I check if my car TE406P is configured as E1's and not T1's ?
18:03.05jake1932flujan: ztcfg -vv
18:03.19jake1932flujan: should show the correct number of channels
18:03.24jake1932flujan: might be other ways
18:03.44*** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr)
18:04.17jake1932flujan: or zttool
18:05.22flujanjake1932, ztcfg -vv show the 62 channels...
18:05.46jake1932flujan: any errors?
18:05.53flujanjake1932, no...
18:06.04jake1932flujan: good
18:06.06flujanjake1932, but and about the output of the zap show status?
18:06.37*** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
18:06.53jake1932flujan: ?
18:07.19flujanjake1932, zap show status shows me three configured spans... but I have only two... :P
18:07.25jake1932:)
18:07.43paolobHi guys! I have an extension "exten => pablocelular,1,Dial(SIP/${CELULARPABLO}@pstn-spa3000-mision,60,Tt)". What's the reason why asterisk doesn't accept a Dial(SIP/pablocelular,60,Tt) ?
18:07.57jake1932flujan: already told you i don't know - answer still remains
18:08.16flujanjake1932, ok... thanks... :D
18:08.25flujanI think I will try the list
18:09.00Peacefulmercestes: does the 7960 have a web interface?  It doesn't seem to respond on port 80.
18:09.22mercestesPeaceful:  yes...it does.
18:09.53*** join/#asterisk Hymie (i=hymie@L8R.net)
18:10.02Hymiedoes anyone know why calls never timeout in my queue?
18:10.07Hymiethe default of 5 minutes, or any other time, it doesn't matter.  I can leave a call in the queue for hours even
18:10.50jake1932Peaceful: it has a telnet interface
18:11.01jake1932Peaceful: different port
18:11.38jake1932Peaceful: actually, a telnet server, and a minibrowser client
18:11.46jake1932Peaceful: no web server
18:12.37Peacefuljake1932: sooo, I should try browsing to the telnet port?
18:14.59Peacefulhmm..I don't know what mercestes was thinking, but I've never seen any mention of a web interface on a 7960 IP phone, and it's not answering.
18:15.44PeacefulBack to my original question: Can you SET values via the telnet interface?  The docs I've found so far only seem to go as far as letting you simulate button presses, which is a pain in the butt.
18:16.14Strom_CPeaceful: why do you need to set values via the telnet interface in the first place?
18:17.20Dr-Linuxwe are going to buy a new server for our TE210P cards, brand should be DELL or Sun ..
18:17.26Dr-Linuxany recommendations?
18:17.48gmfmHymie: when you use the Queue app, option 'n' will prevent it from retrying on timeout and go to the next step in the dialplan
18:17.49PeacefulBecause the phone is 10 miles away and I don't want to keep rebooting the thing over and over just to get it to re-read the tftp config so that I can test settings
18:18.26bzbwStrom_C: tried with priority when using _6xxx for hints, it does not work.
18:18.46Strom_CPeaceful: that's why you test with a phone locally and get it working /before/ you deploy the thing :)
18:18.53diclophiswhy would my sip register attempts be timing out?
18:19.05PeacefulDr-Linux: Opterons
18:19.21Dr-LinuxPeaceful: what's Opterons?
18:19.49Dr-LinuxPeaceful: this server will be running high IVR solutions
18:19.52PeacefulStrom_C: And how do I test the home-user's NAT setup here???   I've got * ->NAT->  Internet ->NAT-> home user cisco
18:20.23Strom_CPeaceful: if you're doing SIP, you should never have more than one NAT in the connection.  SIP does not like to play with NAT.
18:20.44PeacefulDr-Linux: Dell's all use Intel processors for now, Sun sells really expensive AMD opterons.  I was just recommending AMD Opterons over Intel anything.
18:20.45Strom_Chaving the telephone and the asterisk box behind NAT is a recipe for disaster
18:20.58PeacefulStrom_C: So I've noticed.
18:20.59Strom_Cor, more accurately, behind different NATs
18:21.13PeacefulStrom_C: Signalling works fine, I'm just not getting voice
18:21.32Dr-LinuxPeaceful: can you suggest me any from Sun then?
18:21.37Strom_CPeaceful: yes, thats the problem
18:21.42Strom_Cit's a NAT issue
18:21.43vijatiti had asked this question an hour back, since the channel is active now, i think i'll ask it again to try my luck
18:21.46PeacefulStrom_C: which is annoying, since they both seem to be UDP ports
18:21.54Strom_Cno amount of twiddling with the phone will get it working
18:22.02Strom_Cfix your broken network setup
18:22.07PeacefulDr-Linux: If you've got the money, yes.
18:22.33Hmmhesaysthe tick tock of the clock is painful, all sane and logical, I want to tear it off the wall
18:22.37Dr-LinuxPeaceful: what about X4100 from Sun?
18:23.09vijatithas anyone tried iax with custom packetization duration. (In sip, this is possible with "packetization=" directive ).
18:23.16Strom_CHmmhesays: you're only allowed to sing in the channel if you want to hear me sing "take on me" eight hours per day for the rest of your life :)
18:23.53PeacefulDr-Linux: I've never been able to afford a Sun, so I can't recommend any specifics.  I highly recommend NOT going with the XEONs from Dell or anyone else.  Maybe some others here could give better advice on which Sun model would be best.
18:24.06Dr-Linuxanybody likes DELL servers? :)
18:24.12Delta239me
18:24.21Delta239:D
18:24.23*** join/#asterisk mtaht4 (n=m@reserve-64-79-114-30.wiline.com)
18:24.31Strom_CDr-Linux: I have a TE4xxP working in a Dell Poweredge server
18:24.31salviaduddell rocks da house yo
18:24.45Dr-LinuxDelta239: you like DELL over others?
18:24.51Delta239i don't have one but i worked for Dell for years
18:24.54Dr-LinuxStrom_C: what's the server model?
18:25.06Strom_CPoweredge something-or-other :)
18:25.17Dr-Linuxguys recommend me any DELL server
18:25.17Strom_Cdont remember
18:25.21Strom_Cit's at a client's prem
18:25.24*** join/#asterisk nagl (n=nagl@86.59.54.237)
18:25.27Dr-LinuxStrom_C: i need model number
18:25.31Delta239the poweredge
18:25.37Hmmhesaysand boobies
18:25.42Hmmhesaysapparently there is a shortage in pakistan
18:25.46Strom_Cyes, cant forget the boobies
18:26.12Dr-Linuxwe have more then 70 DELL poweredge servers and 30 Sun
18:26.28Dr-Linuxbut we need buy a new one for Asterisk/TE210P
18:26.32Strom_CI think Hmmhesays was talking about a shortage of boobies
18:26.46Dr-Linuxso need recommended models
18:27.03Dr-LinuxHmmhesays: how many boobies? :)
18:27.17protocoldougi just installed the zaptel drivers (to use ztdummy) but when i issue "zap show channels" * says: "no such command zap", any ideas?
18:27.18Hmmhesayspreferably an even number
18:27.39Dr-Linuxlolz
18:27.59Dr-LinuxHmmhesays: 32 bits boobies? :P
18:28.03_Paulo_protocoldoug, try "load chan_zap.so"
18:28.06Hmmhesaysyeah 31 is bad, parity error
18:28.35Dr-LinuxHmmhesays: 36 also errors but, it's cool :P
18:28.58protocoldoug_Paulo_, that is pointing me in the right direction :) that gives an error, something is missing, many thanks definitely gets me pointed in the right direction :)
18:29.13*** join/#asterisk wasabi (n=wasabi@ubuntu/member/wasabi)
18:29.13Dr-Linuxhhm.. so no recommendation? :S
18:29.20_Paulo_protocoldoug, you are welcome
18:29.27wasabiThere any graphical IVR/call center stuff available for asterisk yet? Commercial even.
18:29.49*** join/#asterisk Mw3 (i=mw3@national.t-error.hu)
18:32.55_Paulo_wasabi, writing one seems to be an intersting project.
18:33.14wasabiWhy do you say that?
18:38.04_Paulo_wasabi, because I like to program for fun.
18:38.07Dr-Linuxhttp://www.dell.com/downloads/global/products/pedge/en/2850_specs.pdf
18:38.12wasabik...
18:38.12salviadudim gooooooing home, bye dudes
18:38.14Dr-Linuxhows this server?
18:40.03*** join/#asterisk gursikh (n=FreePBX9@adsl-68-92-36-133.dsl.hstntx.swbell.net)
18:40.25bzbwlooks like i can't use "_6XXX, hint, SIP/${EXTEN}", this is painful for 100 extensions:(
18:40.37Ariel_argh  a quick question about aastra 480i phones.  I am trying to get one to work with an asterisk box. But the box is on the outside network the phone is behind a nat.  does anyone know how to setup the nat setting on the phones
18:45.40Hmmhesaysshaking that ass on the floor
18:45.41*** join/#asterisk dapatrick (n=dapatric@static-151-204-184-67.pskn.east.verizon.net)
18:45.46Hmmhesaysbumpin and grinding that pole
18:46.03*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
18:46.08Strom_Ctaaaake ooooon meeeeeeee
18:48.26*** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk)
18:48.26Hymiegmfm: option 'n' is only to be used for when you want it to go through a only one retry loop... but you're supposed to be able to set a time (like 5 minutes, and that's the default) for it to go on to the next step of the dialplan...
18:48.48*** part/#asterisk Coyotee (n=root@sipx.ica.net)
18:49.43*** part/#asterisk TonyM (n=TonyM@softins.claranet.co.uk)
18:54.31sevardStrom_C: Take me onnn
18:54.43sevardTAKKKEEEE MEEEEE ONNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNN
18:55.00Strom_CI'll bee gooooone
18:55.12Strom_Chahaha
18:55.36Hmmhesaysanyone know what the name of that new korn song is?
18:55.38sevardit's scientifically proven to get that song stuck in your head for weeks at a shot
18:55.46Strom_Cwow, this is timing
18:55.50Strom_Cthey just called
18:56.22Hmmhesaysahyone?
18:56.24Hmmhesays*anyone
18:56.51sevardStrom_C: you'd probably be the one to have it, know of a telco 1004hz 0db test number?
18:57.19*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
18:57.42[TK]D-Fenderbzbw : not that bad...
18:58.40*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
19:02.53Strom_Csevard: it varies from telco to telco
19:03.00*** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do)
19:03.03*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
19:04.25*** join/#asterisk rstrit (n=rstrit@204.238.218.130)
19:04.34sevardStrom_C: but not a ..general test number?
19:06.50*** join/#asterisk mog_work (n=mogorman@gateway.digium.com)
19:07.45bzbwD-Fender: why, I hate to manage those long list of extensions.
19:08.13Hymiedoes anyone know why calls never timeout in my queue?
19:08.17Hymiethe default of 5 minutes, or any other time, it doesn't matter.  I can leave a call in the queue for hours even
19:09.23*** join/#asterisk sandra78 (n=aerae@200.106.96.110)
19:09.29sandra78help!!!
19:09.30*** join/#asterisk lorinc (n=ang@caracas-4553.adsl.interware.hu)
19:09.42sandra78:S
19:09.48_Paulo_what is your problem, sandra78?
19:10.09sevardwow
19:10.12sevardthis guy on the phone
19:10.18sevardjust tried to teach me how to use wget
19:10.26sevardseriously.
19:11.01Peacefulso...he's like "type w-g-e-t", or what?
19:11.15*** join/#asterisk lorinc (n=ang@caracas-4553.adsl.interware.hu)
19:11.18sandra78<_Paulo_> Does anybody knows how to change the asterisk ringing song? i want to record my own ringing song
19:11.21sevardpeople need to pay attention, if your customer is talking about doing some crazy hacker shit with linux you don't try to teach him how to use wget
19:11.43sevardPeaceful: much more indepth than that, he talked to me for a good 10 minutes while I was trying to shut him up nicely.  I had to use some language
19:11.52_Paulo_sandra78, you mean, music on hold?
19:12.01sandra78no, i mean the false ringing
19:12.06sevardsandra78: do you mean music on hold or indication
19:12.46sandra78when you call an extension you can hear a ring
19:12.47sevardindication.
19:12.47sevardlook at the wiki
19:12.57sandra78i want to chenge this song for another song similar to pstn  song
19:13.03Hymiehum.. is anyone here even using queues?
19:13.14sevardHymie: I am
19:13.29[TK]D-Fenderbzbw :its a 1-shot job mostly.... pretty quick to make...
19:13.40_Paulo_sandra, are you in Brazil?
19:13.47HymieFOR THE LOVE OF GOD MAN, any ideas? ;)
19:13.56sevardwhat's the porblem?
19:14.03Hymiemy queues never timeout
19:14.11Hymiepeople sit in them for 2000 years, if they please
19:14.12[TK]D-FenderHymie : Paste the line you use to call the queue
19:14.13sevardwhats your syntax
19:14.14brad_msswsevard: hopefully you didn't tell him you were 'trying to do some crazy hacker shit' .... he probably immediately took you as an idiot at that point ... hence the reaction
19:14.37sevardbrad_mssw: no, i'm flipping bits and shit, i talk more professional on the phone :)
19:14.42Hymieexten => s,2,Queue(tech-queue|t||30)
19:14.54Hymie30 is a test, I've tried realistic numbers, like 300, etc
19:15.19_Paulo_sandra78, are you in Brazil?
19:15.32Dr-Linuxguys, please confirm me, if this server's slots/buses are compatible with Digium TE210P ? http://www.dell.com/downloads/global/products/pedge/en/2850_specs.pdf
19:15.41*** join/#asterisk jtoy (n=toy@cust-206-40-173-219.bos-static.gis.net)
19:15.43*** part/#asterisk jtoy (n=toy@cust-206-40-173-219.bos-static.gis.net)
19:15.44sevardHymie: my queue is exten => s,4,Queue(co_queue|tT|||300)
19:15.55Hymiesevard: it just never, ever times out here
19:16.02Hymiesevard: what version of asterisk?
19:16.03Dr-LinuxThree total: three PCI-X® slots (64-bit/133MHz) or two PCI Express™
19:16.03Dr-Linuxslots (1 x 4 lane and 1 x 8 lane) and one PCI-X slot (64-bit/100MHz)
19:16.06sevardHymie: try my syntax
19:16.13[TK]D-FenderHymie : You are missing a "|"
19:16.17sevard1.2.7.1
19:16.30Hymiehmm.. ok, I will try this, damned ||| intead of ||
19:16.30[TK]D-Fenderexten => s,2,Queue(tech-queue|t|||30) <- this is right
19:16.47[TK]D-FenderHymie : though you'll want to change that 30....
19:16.56Hymie[TK]D-Fender: sure.. I want 600 anyhow
19:16.56sevardHymie: most problems are silly syntax problems
19:16.59Dr-Linuxput another T to make your life more easy :)
19:17.03Hymie[TK]D-Fender: just had to try everything
19:17.06[TK]D-FenderHymie : You should have been ...RTFM!
19:17.19Hymiewhy another T?  I don't want the caller to transfer anywhere, except to one digit extensions
19:17.19[TK]D-FenderHymie : "show application queue"
19:17.40Dr-Linuxanybody get a change to look into the link that i pasted?
19:17.43sevardI believe the T is queue out, but I don't recall
19:17.46[TK]D-FenderHymie : I didn't add the "t" I assumed you WANTED it there so I LEFT it in from your line.
19:18.10sevard[TK]D-Fender: you LOVE to do THIS a LOT
19:18.18[TK]D-FenderHymie : "tT" are worthless if you're using real phones...
19:18.20Hymie[TK]D-Fender: no, heh.. dr-linux said I shoudl add a T.. that's inocming transfer, according to the wiki.. I want the callee to transfer, but not the dude that called in
19:18.25sevard[TK]D-Fender: at least you don't do this
19:18.35[TK]D-Fendersevard : It compensates for lack of volume control :)
19:18.41sevard[TK]D-Fender: :|
19:19.18[TK]D-FenderHymie : What kind of phones are you using?
19:19.25HymieUniden 200's
19:19.29Hymiequite nice actually
19:19.30sevard[TK]D-Fender: worthless? refresh my memory.
19:19.32HymieI like them
19:19.38Hymiealthough they always clip at the start of the call :/
19:19.41Hymieeven with latest firmware
19:19.44[TK]D-FenderHymie : My condolences... get rid of the "t" you don't need it, they have SIP transfer capability.
19:20.04Hymieno big or small t?  we can't have any 't'?  but I like t ;)
19:20.04[TK]D-FenderHymie : I regret having bougth the 2 that I did....
19:20.09HymieI have a poster of Mr. T on the wall
19:20.13Hymie[TK]D-Fender: why?
19:20.14sevardmmm tea
19:20.14[TK]D-FenderHymie : Niether.  remove it completely.
19:20.22Hymie[TK]D-Fender: what don't you like?
19:20.24[TK]D-FenderHymie : Why what?  My purchase regrets?
19:20.28Hymieyeah
19:20.41*** join/#asterisk flujan (n=flujan@internet.nube.com.br)
19:20.47Hymieah, jolly good, thanks for the | assist, it's fine now
19:20.53[TK]D-FenderHymie : Feels friggen cheap, shitty interface, inflexible provisioning, poor button placement, etc
19:20.54Hymiehalf a day wasted on that
19:20.58sevardI regret buying a PAP2 thinking I might be able to hax it
19:21.15Hymie[TK]D-Fender: sure, they feel cheap, but they ARE cheap too ;)
19:21.17[TK]D-FenderHymie : Polycom IP 301 is worth every extra penny.
19:21.22sevardHymie: google, site:voip-info.org <keyword>, saves time.
19:21.26Hymie[TK]D-Fender: well, tell that to my client ;)
19:21.30Hmmhesaysi've never played with the 301's
19:21.37Hymiesevard: ?  I've been there, I just didn't notice the extra | missing
19:21.50Hymiesevard: one of those things... it just happens sometimes
19:21.51sevardglasses?
19:21.57sevarddyslexia?
19:22.00[TK]D-FenderHymie : thats what I bought those 2 for vs the 26 * Polycom IP600's for the office.  those 2 are in "high-rape-risk" areas so I wouldn't feel a loss if something unfortunate were to happen to them.
19:22.00sevardi'm dyslexic as fuck
19:22.03Hymiesevard: ah, I'll make sure to hand you a Mr. Perfect star later ;)
19:22.14Hymiesevard: I just missed it, I usually don't
19:22.23sevardwhy does this remind me of that movie
19:22.27Hymie[TK]D-Fender: cool
19:22.34flujanHi all, I'm having problems. Every time I try to place a call using asterisk I receive: http://pastebin.com/735639
19:22.40sevardsome guy says some dude told his crazy sister there were ghosts in her clothes
19:22.41[TK]D-FenderHmmhesays : The 301 is pretty nice, I use one at home as well as a 501.  I use 60x's at work mostly.
19:22.42sevarddamn.
19:22.52[TK]D-FenderHmmhesays : the new IP 430 is going to severly ROCK....
19:22.53HmmhesaysI have a snom 190 at home
19:22.53Hymiehehe
19:23.16[TK]D-FenderHmmhesays : A category killer I'm sure....
19:23.26Hymieanyhow, thanks guys.. off to finish this build
19:23.28Hmmhesayssay what?
19:23.42[TK]D-FenderHmmhesays : The new Polycom IP 430....
19:23.43Hmmhesayswhats up with the 430? looks like a 501
19:24.00[TK]D-FenderHmmhesays : nope, its built on a 301 fram, but with MASSIVE improvements
19:24.08[TK]D-Fenderframe*
19:24.09Hmmhesayssuch as
19:24.14sevardoh man, this jerky doesn't taste hot
19:24.17sevardbut when it gets in your stomach
19:24.21sevardyou regret eating it
19:24.27*** part/#asterisk Hymie (i=hymie@L8R.net)
19:24.43tdonahue-laptopthanks sevard, just a little too much information though...
19:24.48[TK]D-FenderHmmhesays : Full pixel LCD, 4 soft keyes, arrorw key navigation, PoE INTEGRATED, LED's for the line keys, speakerphone and more.
19:24.57sandra78<_Paulo_> Hi you mean do i have to change in the indication.conf file?
19:25.00sevardtdonahue-laptop: i do NOT look forward to that coming out of my brown star
19:25.01[TK]D-FenderHmmhesays : http://www.polycom.com/products_services/0,1443,pw-34-182-15672,00.html
19:25.08*** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon)
19:25.09_Paulo_sandra78, sure...
19:25.27Hmmhesayswhats the price on these?
19:25.40Hmmhesaysyeah i was reading that [TK]D-Fender
19:25.46sevardthose phones look like crap
19:25.55_Paulo_sandra78, you can change the tones there.
19:25.57[TK]D-FenderHmmhesays : Its slated to fit betweent he 301 & 501, so it may push each a little or just push the 301 down I suspect.
19:26.04sevardthe Aastra 480i already has that AND more
19:26.27*** join/#asterisk nialp (n=nialp@217-162-135-208.dclient.hispeed.ch)
19:26.30sevardwayyyyyyyyyyyyyyy more
19:26.33Hmmhesaysi have yet to find a better speakerphone than polycom
19:26.36[TK]D-FenderHmmhesays : with 301 @ $115 and 501 @ $170 that tells me about $140 which would be a killer.
19:26.45_Paulo_sandra78, there is a country=your_country_code in this file
19:26.46sevardaastra :)
19:26.50Hmmhesayswhats the speakerphone like on teh 301?
19:27.11[TK]D-FenderHmmhesays : NONEXISTANT :)
19:27.22[TK]D-FenderHmmhesays : but he SPEAKER is just like the others :)
19:27.24Hmmhesaysoh...  that out for any of my installs then
19:27.36nialpafter nearly 1 year of service my asterisk with passiv HFC-S in NT mode doesn't supply a dial tone any more
19:27.51sandra78exten => 264,3,Playtones(!950/330,!1400/330,!1800/330,0)
19:27.52nialpi upgraded to asterisk 1.2
19:28.02nialpand still no dial tone
19:28.27Hmmhesaysthat suggests hardware failure
19:28.27nialpi can dial out and every thing seems to work, but the dial tone
19:28.32_Paulo_sandra78, look at that line and see if it matches your country.
19:28.37[TK]D-FenderHmmhesays : Thats why the 430 is so disruptive.  the 501 didn't have lights, the 301 & 501  didn't have proper integrated PoE, and this ones price point fits betwwen both.  a class of its own.  It practically replaces the 301 & 501 simultaneously in my mind
19:28.58Hmmhesayswhen is it coming out?
19:29.03[TK]D-FenderHmmhesays : About a month
19:29.15[TK]D-FenderHmmhesays: I am SO gonna pawn off my IP 301 for it :)
19:29.24Hmmhesaysi'll get a contract to buy me one
19:29.27[TK]D-FenderHmmhesays : with any luck AT PAR :)
19:29.43Qwell[][TK]D-Fender: got a link?
19:29.48Hmmhesays[TK]D-Fender have you ever played with openwrt?
19:29.49[TK]D-FenderQwell : see above
19:30.00sandra78i was using exten => 264,3,r
19:30.04Qwell[]gotcha
19:30.18*** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com)
19:30.25[TK]D-FenderHmmhesays : Nope, I have the best grade router to choose for it but never got around to playing with it.
19:30.33Qwell[]I'll wait for the 630
19:30.40vooduhalHey guys.  Can anyone point me in the direction of a MIB for the Polycom IP phones?
19:30.47Hmmhesaysi have an audiocodes ac494 board sitting here
19:30.50Hmmhesaysrunning linux
19:30.55[TK]D-FenderQwell : 630?  Fictitious?  Or linkable?
19:30.57Hmmhesaysfxs and fxo ports
19:31.06Qwell[][TK]D-Fender: fictitious
19:31.17Qwell[]this has no mini browser
19:31.19DaminHow much is the 430 going to be selling for retail?
19:31.19Qwell[]lame!
19:31.21[TK]D-FenderQwell : thats for gettin my hopes up!
19:31.30Qwell[]Damin: I don't think it's been stated
19:31.38[TK]D-FenderDamin : As I'm told, in between the 301 & 501
19:31.42Qwell[][TK]D-Fender: What would you guess though, $140?
19:31.52FlautoMay 24 14:31:09 WARNING[17213]: app_dial.c:1162 dial_exec_full: Invalid timeout specified: '+asterisk'
19:31.55[TK]D-FenderQwell : roughly in my estimate.
19:31.58Flautowhat is that
19:32.12Damin[TK]D-Fender: My scrollback is fucked..
19:32.16Qwell[]http://www.polycom.com/products_services/0,1443,pw-34-182-15672,00.html
19:32.25[TK]D-Fender[15:26] <[TK]D-Fender> Hmmhesays : with 301 @ $115 and 501 @ $170 that tells me about $140 which would be a killer.
19:32.26Qwell[]Damin: ^
19:32.44Qwell[][TK]D-Fender: heh, I didn't even see that line...I guessed
19:32.53[TK]D-FenderFor full PoE + Brick, and all the other plusses it looks kinda sick.
19:33.10[TK]D-FenderQwell : Both fair guess' :)
19:33.18flujanhttp://pastebin.com/735639 someone have a idea why this is happening?
19:33.39flujani'm in trying this in my loopback environment.
19:34.36JackEStormDamin: look it up on froogle
19:34.50Qwell[]JackEStorm: it helps if the product existed
19:34.59vooduhalAlso, does anyone know of decent mib viewer for *nix?
19:36.25[TK]D-Fendervooduhal : What is this about MIB?
19:36.56vooduhalJust needing to know if there a polycom specific MIB for the IP phones, specifically 500 and 600s.
19:37.05[TK]D-FenderDamin, Hmmhesays, Qwell : Also integral to the release of the 430 is SIP 2.0 firmware. <-
19:37.17Qwell[]2.0, pfft
19:37.22Qwell[]cisco is already on 8.x
19:37.22[TK]D-Fendervooduhal : Can you please clarify your term "MIB"?
19:37.28Qwell[]so far behind
19:37.31vooduhalSNMP mib database.
19:37.32DaminDAMN!!!! And you say that is going to be about $140 retail?
19:37.38[TK]D-FenderQwell : With Cisco, 20th time's the charm ;)
19:37.41Qwell[]:p
19:37.51[TK]D-FenderDamin : Wicked cool ain't it? :D
19:37.54Qwell[]add a factor of 10, and you'd be pretty close
19:38.01Damin[tk] Yep..
19:38.22[TK]D-FenderQwell : They're using DECIMAL revisions to get there.. it may as well be a factor of ten ;)
19:38.24*** join/#asterisk sb_mx (n=sb_mx@200.94.154.226)
19:38.32[TK]D-FenderDamin : Category killer stuff....
19:38.56[TK]D-FenderDamin : No normal user could need more than that phone.
19:39.05[TK]D-FenderDamin : And at Polycom quality.
19:39.09Daminstkn: Yep..
19:39.24Damin[TK]D-Fender: Yep.. it's my new favorite phone. ;)
19:39.38DaminAnyone know if the Polycom Attendant Console works w/ asterisk? http://www.polycom.com/products_services/0,1443,pw-34-182-12072,00.html
19:39.47[TK]D-FenderDamin : Rest assured I will get one immediately upon its release.
19:39.47Qwell[]but no minibrowser...bah
19:39.50Qwell[]I'm not thrilled
19:40.02[TK]D-FenderDamin : Works great, I've got 2 fully loaded for my receptionist
19:40.05vooduhalSo no ideas?
19:40.09DaminI've got a live-answer office application and that thing just looks awesome! :)
19:40.13[TK]D-FenderQwell : And how many people seriously care about that?
19:40.24CunningPikeDamin: afaik, the Attendant Console is just a 601 and a bunch of sidecars........
19:40.52Qwell[]1 at least
19:41.02Qwell[]coming from cisco, it's obligitory :p
19:41.12SplasPoodHeh, anyone try the new cisco 7960 firmware 8.3 yet?
19:41.15[TK]D-FenderCunningPike : Actually the term applies to the sidecar alone.
19:41.45CunningPike[TK]D-Fender: Didn't know they worked by themselves.......
19:41.48SplasPoodDon't
19:41.52SplasPoodif you haven;t
19:41.56SplasPoodit seems to have.. issues
19:42.05Qwell[]SplasPood: so don't use sip
19:42.09C4T3ldoes anyone know what the newest sip.ld version is on the polycom 601?
19:42.09SplasPoodDamin: Yes, it does
19:42.14Qwell[]I'll be the skinny firmware works great. ;)
19:42.17SplasPoodDamin: the polycom.. as of firmware 1.6.6
19:42.20Qwell[]bet*
19:42.23[TK]D-FenderCunningPike : they don't just clarifying the term because the 601 is a phone in its own right, the module use iin conjunction would be for an attendent.
19:42.24*** join/#asterisk zotz (n=zotz@24.231.36.9)
19:42.31SplasPoodDamin: it always worked before, but was limited to 7 monitored extens
19:42.41SplasPoodC4T3l: 1.6.6
19:43.10*** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com)
19:43.21C4T3lSplasPood: thanks. I'm trying to get presence working with the extension module
19:43.26[TK]D-FenderSplasPood : No longer limited (for practical amounts) at this point.  you can have a phone + 3 modules (the limit) fully loaded now
19:43.44[TK]D-FenderC4T3l : it works very well here.
19:43.47*** join/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net)
19:43.49SplasPoodC4T3l: it'll work, but only 8 people on the 500, and only /w firmware 1.6.6 .. before that it was 7 people in all places
19:43.55TripleFFFFFFFFFFhey
19:43.58Hmmhesaysi love that episode of voyager
19:44.08SplasPood[TK]D-Fender: See above.. I already said that to Damin
19:44.09DaminCool.. so we'll be able to set status on phones and the attendant will visually see if a person is available or not?
19:44.13TripleFFFFFFFFFFis there a way to force a codec in the dialplan for  a certain NPA ?
19:44.16SplasPoodDamin: yup!
19:44.19[TK]D-FenderI had them since december without proper functionality.  * adding SIP-B support for 1.4 will make them that much more powerful.
19:44.22SplasPoodup to 48 hints now
19:44.23C4T3lSplasPood: thanks. how many with the 601?
19:44.34Assid1.6.6 isnt public yet
19:44.37SplasPoodC4T3l: 48
19:44.41SplasPoodAssid: yes it is
19:44.46Assidit is ?!?!?
19:44.52SplasPoodAssid: its just not OLD so its not on the site /wo access
19:44.57C4T3lSplasPood: oh, with multiple mods
19:45.09SplasPoodC4T3l: yes, or via the on screen buddy list
19:45.32TripleFFFFFFFFFF??
19:45.33Assidi have access.. but i dont think they gave me enough access to download sip updates
19:45.38[TK]D-FenderSplasPood : Yeah, but the buddy list browser is a totally ass way of doing it...
19:45.51SplasPood[TK]D-Fender: true, but it is possible :)
19:46.03C4T3lwhere could i get 1.6.6
19:46.08Qwell[]C4T3l: Your reseller
19:46.13SplasPoodC4T3l: Who'd you bu.. what Qwell said
19:46.15[TK]D-FenderSplasPood : A handset rectal exam is possible but I wouldn't suggest it ;)
19:46.52TripleFFFFFFFFFFso i guess its not possible ?
19:46.53C4T3lSplasPood: not sure. my boss just kinda threw it at me and said make it work
19:47.46AssidSplasPood: i go to voice downloads..
19:47.49Assidbut no file :(
19:48.04TripleFFFFFFFFFFyou r sure no file ?
19:48.09TripleFFFFFFFFFF;)
19:48.14Assidnah
19:48.25TripleFFFFFFFFFFhehe s there a way to force a codec in the dialplan for  a certain NPA ? someone ?
19:49.04Assidcan i grab 1.6.6 of someone here?
19:49.16C4T3lme too
19:49.27C4T3lor is that a violation of some law?
19:49.41*** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk)
19:50.31SplasPoodYea its totally not legit, I believe
19:50.43SplasPoodAlthough on the other hand I don't see why it SHOULD be..
19:51.40sb_mxevening everyone, i have a question.. is there a specific reason why we have to reload everything when we make a change to a context or a specific extension? ie: extension 123 now uses context from-int. i'd like to only reload the "information" associated with that extension
19:51.59sb_mxwe've been toying around with chan_sip in order to prevent this
19:52.24sb_mxbut we're not sure it  a)  is stable or b) makes sense
19:52.27SplasPoodsb_mx: whats wrong with reloading all the extensions?
19:53.11sb_mxSplasPood, when you have more than 100 extensions with different contexts, it takes a lot of time to reload all the extensions/globals
19:53.25SplasPoodI suppose
19:54.04sb_mxso we were actually thinkin of doing something like "sip reload 123" instead of just doing "sip reload" from the cli
19:54.29SplasPoodhow would you deal with a removal?
19:54.34flujansomeone already configure a mfc/r2 E1?
19:54.57sb_mxSplasPood, in that case you'll have to do sip reload
19:55.04TripleFFFFFFFFFFas well as when you have 50,000 sip entries.. sip reload is nasty
19:55.16SplasPoodrealtime?
19:55.54SplasPoodTripleFFFFFFFFFF: and whyy would one scale to 50k sip.conf entries on 1 box?
19:56.16sb_mxwell, according to realtime's documentation, whenever you want to update something you have to reload the configuration so everything gets flushed
19:56.25*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
19:57.17TripleFFFFFFFFFFrealtime
19:57.54TripleFFFFFFFFFF100 clusterd boxes could have 50,000 entries shared
19:57.58TripleFFFFFFFFFFor not so shared
19:58.33TripleFFFFFFFFFFi think theres a big overhead in the qualify statement etc.. when over 300 entries
19:58.51TripleFFFFFFFFFFlots of crap going on.. but that my 0.00009 per min toughts
20:01.47sb_mxanyways, do you think its reasonable to make a single extension reload when you're only changing that one?
20:05.07*** part/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net)
20:06.41achandrahello.alot of peeps are asleep on the ser channel. Does anyone have direct experience with dispatcher module in version 1.10 of ser to get asterisk LB AND deal with failover?
20:07.06*** join/#asterisk _alex_mx (n=_alex_mx@200.94.154.226)
20:09.09*** join/#asterisk MattH (n=MattH@63.174.244.195)
20:09.16MattHHi... is thereanyway I can 'spoof' a BLF to a phone?
20:09.23MattHthat being can I send a BLF notification (on or off) from the dialplan?
20:10.12Hmmhesaysanyone know how to do a port range in iptables
20:10.23Hmmhesaysi don't remember
20:10.25docelmocarefully
20:10.31*** join/#asterisk ikey (n=er@203.115.29.2)
20:10.34Hmmhesaysyeah no doubt
20:10.46Hmmhesaysnow for a useful answer
20:10.49docelmoI did it once and ended up having to reinstall linux
20:10.49Hmmhesays...
20:10.50Qwell[]:
20:10.56Qwell[]I thought
20:11.05Hmmhesaysi meant forward port ranges
20:11.11Qwell[]10:20
20:12.39*** join/#asterisk TUplink (n=Tommy@68-232-82-147.chvlva.adelphia.net)
20:12.52TUplinkcan you sent instant messages over SIP with asterisk
20:14.15fileTUplink: hello to you too
20:14.30C4T3liptables -I FORWARD -d <ip> --dport 10:1000 -j KILLBOSS
20:14.49achandralol
20:15.05*** part/#asterisk sb_mx (n=sb_mx@200.94.154.226)
20:15.12*** join/#asterisk sb_mx (n=sb_mx@200.94.154.226)
20:15.16achandrathe KILLBOSS parameter is the best.
20:15.25*** part/#asterisk sb_mx (n=sb_mx@200.94.154.226)
20:15.27*** part/#asterisk diclophis (n=diclophi@65.203.37.58)
20:15.35*** join/#asterisk sb_mx (n=sb_mx@200.94.154.226)
20:15.35C4T3lit never seems to work tho :(
20:16.12*** join/#asterisk adorah (n=Asterjet@87.69.72.228)
20:22.36*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
20:23.41*** join/#asterisk slobberknocker (n=ckwall@63.149.122.94)
20:24.57slobberknockeris there an equivalent for voicemail.conf like extensions reload and sip reload... something to the effect of voicemail reload? I cant figure out what command works.
20:26.12dlynes_officeslobberknocker: yeah...reload
20:26.19dlynes_officeslobberknocker: or reload app_voicemail.so
20:26.38slobberknockerand that will not affect active channels or anything?
20:26.53dlynes_officeslobberknocker: reload app_voicemail.so though does a complete reload; it doesn't just reload the config file
20:27.03sb_mxslobberknocker, also, you should try typing 'help' from the cli that'll give you some commands and their explanation
20:27.35dlynes_officeslobberknocker: reload app_voicemail.so will probably affect current channels that are using voicemail. yes
20:27.37slobberknockeryeah, I was reading those... i was looking for something that would do just voicemail without affecting 10 users that are in a conference call right now.
20:27.48*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
20:28.01dlynes_officeslobberknocker: what does the conference call have to do with voicemail?
20:28.09slobberknockernothing...
20:28.18*** part/#asterisk terrapen (n=cjs@166.70.183.108)
20:28.19slobberknockeri am making changes to voicemail and need to reload the config
20:28.22slobberknockerbut i have users on calls
20:28.28dlynes_officeslobberknocker: Just use reload
20:28.43dlynes_officeslobberknocker: Then you don't have to worry about whether anyone's currently logged into voicemail or not
20:28.47achandrathat wont kick users?
20:28.51dlynes_officeachandra: no
20:28.52slobberknockersweet!!! thanks
20:28.54sb_mxslobberknocker, doing a reload or relaod app_voicemail.so shouldnt wont current channels. the channels will be "affected" from the next call on
20:29.09slobberknockerok
20:29.10sb_mxslobberknocker, wont affect
20:29.15achandragood to know..
20:29.17slobberknockerthanks
20:29.26achandra;) good question
20:29.39slobberknockerits about time i dont as a stupid one
20:29.45dlynes_officesb_mx: reload app_voicemail.so will affect current channels that are accessing voicemail though, won't it?
20:29.51slobberknockerstupid 10 not stupid 1
20:29.52*** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252)
20:30.52*** join/#asterisk IceManRISK (n=kart@200.138.71.129)
20:31.07*** join/#asterisk postel_ (n=jp@unaffiliated/postel)
20:31.48sb_mxdlynes_office, if i recall correctly, they wont. i think asterisk "locks" those channels. not sure if its the same, but when you switch a context for an extension, it will start working as soon as the agent hangs up and calls again
20:32.01*** join/#asterisk supjigatr (n=syslod@152.53.16.10)
20:32.33dlynes_officesb_mx: ah...so it's the same as restart when convenient, but applicable to that module only?
20:32.36*** join/#asterisk loonacy (n=loonacy@24-117-254-250.cpe.cableone.net)
20:33.31sb_mxdlynes_office, i think so, yes. although im not one of the coders so whatever i tell you is from hand-on experience only
20:34.06dlynes_officesb_mx: yeah...well, that's how i gained knowledge of the reload command...watching it, and realizing that it didn't boot peeps :)
20:34.08supjigatrAny seen any virtual PBX dialplan examples, docs or ideas?
20:34.32dlynes_officesb_mx: it actually issues a notice to your screen to let you know that there was active calls on certain channels or whatever, when you do a reload
20:34.44docelmoOI!
20:34.49docelmoerr somethin
20:35.02sb_mxdlynes_office, yup and it'll "reload" those channels as soon as the lock expires
20:35.22*** part/#asterisk Peaceful (n=Peaceful@70.98.162.62)
20:35.23dlynes_officesupjigatr: what exactly are you trying to do?
20:36.08supjigatrWell are working on verison 2 of our virtual PBX and I'm not happy with some of what we did.  Basically we are hosting Virtual PBX's.
20:36.53supjigatrI'm looking for a good model to follow or just bounce some ideas around.
20:37.24dlynes_officeah
20:38.01Delta239any of you running a predictive or an autodialer?
20:38.12s0lidsupjigatr, so what's on your mind
20:38.22*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
20:38.30*** join/#asterisk sulan (n=ksjoberg@82.182.83.84)
20:38.40dlynes_officeDelta239: you're looking for documentation on it?
20:38.58Delta239yeah.. if you have some please
20:39.09dlynes_officeDelta239: do a search on voip-info.org for 'call file'
20:39.46*** join/#asterisk somegeek (i=levin@unaffiliated/somegeek)
20:39.49supjigatrs0lid: well handling multi MOH, call limits etc.
20:40.16Delta239lets see
20:40.54sulanHow come Asterisk only finds sip users/peers by the username in From-header, ignoring the domain?  Doesn't that cause problems when handling incoming calls from SIP-users on a foreign domain when usernames clash?
20:41.28supjigatrS0lid: Are you doing virtual PBX?
20:42.36s0lidnot exactly PBX
20:42.43s0lidwe are running inbound and outbound call centers
20:42.43*** part/#asterisk slobberknocker (n=ckwall@63.149.122.94)
20:43.02supjigatrAh.
20:43.18supjigatrWe have a large call center 480 phones.
20:43.37supjigatrBut we also have been dabbling in the small 4-6 phone virtual PBX.
20:43.53supjigatrI'm working on flowcharting out a good dialplan design.
20:46.35Netgeeks.
20:46.42asterboyI need help hooking up this phone to my * box:
20:46.45asterboyhttp://cgi.ebay.com/Vintage-Fisher-Price-chatter-telephone-C-1961_W0QQitemZ6059862027QQcategoryZ374QQrdZ1QQcmdZViewItem
20:47.25asterboyAll it does is move the eyes when I dial. :(
20:47.50*** join/#asterisk tdonahue-laptop (n=tdonahue@64.201.13.51)
20:48.11dlynes_officeasterboy: that's because it's not a real phone
20:48.18asterboydam
20:48.30dlynes_officeasterboy: i used to have one of those when i was a kid :0
20:48.35asterboylol
20:48.49asterboySo much for selling those in the office.
20:49.11dlynes_officeasterboy: well, if you bought it, you would know it's not a real phone, you dork
20:49.31dlynes_officeasterboy: there's no phone cord on it
20:49.52asterboyya I know, just kiddin around.
20:49.56supjigatrHow do I convert this to the new CUT sytax? exten => _9NXXXXXX,1,Cut(channeltype=CHANNEL,/,1)
20:49.58asterboyHey how about this one then!
20:50.01asterboyhttp://cgi.ebay.com/Vintage-Rainbow-Trout-Telephone-for-Den-w-Reels-Rods_W0QQitemZ7243545673QQcategoryZ793QQrdZ1QQcmdZViewItem
20:50.55dlynes_officewhat an uuuuuuuuuugly phone
20:51.15asterboygreat gift for ya special gal
20:51.24dlynes_officeoh yeah definitely
20:51.43dlynes_officei think she'd much rather have that than a nice meal at a fancy restaurant :p
20:52.58Qwell[]supjigatr: Set(channeltype=${CUT(CHANNEL,/,1)})
20:54.12*** join/#asterisk kuku5 (n=kuku5@c-71-201-217-245.hsd1.il.comcast.net)
20:54.22*** join/#asterisk Qwell (n=north@unaffiliated/qwell)
20:54.48docelmoVonage goes public today and they are down over $2 from IPO
20:54.51*** join/#asterisk viLeR (i=1000@200.114.70.228)
20:54.56kuku5For some reason sip isnt going through my firwall ( cisco ), any suggestions on how to filter it to see whast happening ? ( outgoing calls work fine )
20:54.57docelmoTheir stock is gonna tank
20:55.00dlynes_officehahhaaha
20:55.17SpaceBassthats kind of a bad thing for VoIP in general
20:55.25sb_mxkuku5, maybe dnat to your * box?
20:55.59kuku5dnat?
20:56.10kuku5<PROTECTED>
20:56.18kuku5for 5060 - 5063 udp
20:56.43sb_mxkuku5, are you somehow returning the traffic from the firewall to your * box?
20:57.17asterboyWonder if these support SIP?
20:57.19asterboyhttp://cgi.ebay.com/Vintage-Bells-Patent-Telephone-Receiver-Transmitter_W0QQitemZ6631463015QQcategoryZ38037QQrdZ1QQcmdZViewItem
20:57.50*** join/#asterisk boch (n=root@201.216.241.97)
20:58.02*** join/#asterisk ghost99 (n=neville@222-153-178-14.jetstream.xtra.co.nz)
20:58.11SpaceBassasterboy, yeah but you have provide the 1s and 0s
20:58.27asterboylol...talk in binary
20:59.00SpaceBassyou know what they say....there are only 10 kinds of people, those who understand binary and those who dont
20:59.05kuku5sb_mx: Im forwarding from the firewall to *
20:59.08bochwhat is the acceptable latency between an asterisk and a sipura to establish 2 telephone calls ?
20:59.23dlynes_officedocelmo: dood....that totally rocks
20:59.35dlynes_officedocelmo: pretty soon we won't have to compare our rates to vonage
21:00.11docelmoWho?
21:00.16sb_mxkuku5, are you missing sound or traffic to from incoming calls
21:00.18docelmoWho's rates
21:00.22docelmoI never did..
21:00.25asterboyya, I like that joke...but what about the other 9 people? :P
21:00.27docelmoI was always cutting edge
21:00.28dlynes_officedocelmo: anyone's rates
21:00.34kuku5traffic - the call doesnt get to it, broadvoice things its busy
21:00.38dlynes_officedocelmo: you mean same rate as vonage?
21:00.52docelmoNope..  Im MUCH cheaper across the board
21:00.52sulanI just tried placing a call from an unregistered SIP-client with the same username (but other domain) as a registered sip friend at my * box.  I got a 407.. :(
21:00.54Daminboch: Whatever you feel is acceptable. I have some clients that have 600 Ms of latency and although I would never use it, they love it.
21:01.06dlynes_officedocelmo: even for unlimited minutes?
21:01.48docelmoI would have have to check our retail platform
21:02.13dlynes_officedocelmo: yeah...we can beat vonage at everything except the unlimited packages
21:02.21dlynes_officedocelmo: those packages, we can't even come close
21:02.42docelmoIm not sure what were selling for honestly..  Im more BYOD/Wholesale
21:03.12dlynes_officedocelmo: $30/mo unlimited north america
21:03.25dlynes_officedocelmo: $40/mo unlimited north america for businesses
21:03.42dlynes_officedocelmo: and that's $Cdn
21:03.44docelmohmmm brb  Im gonna go have a look see
21:03.48*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
21:04.16docelmo$25USD unlimited US/Canada
21:04.39docelmoWanna buy some A-Z wholesale?
21:04.42docelmo:)
21:04.47sevardhow much to liberia
21:04.58_alex_mxisn't US/Canada in north america?
21:05.05*** join/#asterisk AuPix (n=AuPix@adsl-04-85.abel.net.uk)
21:05.10dlynes_office_alex_mx: yes
21:05.18docelmono
21:05.24_alex_mxso is it 25 or 30
21:05.28dlynes_office_alex_mx: as is mexico
21:05.29sevarddlynes_office: how much for Liberia
21:05.38docelmoMexico is central
21:05.53dlynes_officedocelmo: ah...maybe in American geography
21:06.00sevardMexico is still considered north in most cases
21:06.11_alex_mxuhmmm not, which is why it's called the north american free trade agreement
21:06.11dlynes_officedocelmo: in Canadian geography, we count it as North America
21:06.15docelmowhatever..  I hate geography
21:06.20sevardheh
21:06.21sevardso
21:06.22sevardum
21:06.23sevardLiberia?
21:06.34docelmoWhere the hell is liberia?
21:06.37dlynes_officesevard: we don't do wholesale
21:06.39sevardFUCKING AFRICA
21:06.40sevardjk
21:06.42docelmoohh
21:06.43sevardbut seriously
21:06.45sevardafrica.
21:06.45dlynes_officeNorth Africa
21:06.58*** join/#asterisk aze_ (n=aze@ACayenne-101-1-6-117.w81-248.abo.wanadoo.fr)
21:07.01dlynes_officeBorders the Mediterranean
21:07.02sevarddocelmo: did you say you do A-Z?
21:07.10docelmocheck out http://www.plainvoip.com/?page=showrates
21:07.15sulanam I asking a dumb question?  I have searched voip-info.org and just got it confirmed - but ain't it gonna pose a problem when accepting calls from any given SIP-client on the 'net?
21:07.18sevardi like my voip plain.
21:07.22sevardvanilla
21:07.22docelmoBYOD and wholesale
21:07.41sevardnon-BYOD services are uber-gay
21:07.45sevardi would NEVER buy
21:07.54Hmmhesaysshut up
21:07.55docelmoYou probably dont have the minutes to qualify
21:08.04SpaceBasshey Hmmhesays whats happenin'
21:08.15Hmmhesaysworking with idiots
21:08.16Hmmhesaysyou?
21:08.30SpaceBassjust finished working with idiots
21:08.35HmmhesaysI got really drunk last night and was 1 1/2 late for work
21:08.35dlynes_officesulan: you're accepting connections from any sip client?
21:08.38SpaceBassnot im working up the mental will power to go buy beer
21:08.39docelmoIm leaving my idiots now..
21:08.43docelmocya @ home..
21:09.03sulandlynes_office: well, I thought it would be good if you could call the office without going through PSTN first.. :)
21:09.04sevardi'm having major pipe issues visiting plainvoip.com
21:09.20dlynes_officesulan: use a username and password to access it
21:09.29dlynes_officesulan: otherwise you're asking for trouble
21:09.56dlynes_officesulan: also force them to do sip registrations
21:10.23sulandlynes_office: all office extensions does... but for example, if an office extension has the username foo and a sip client on the net with the local username foo tries to place an INVITE to my * box it replies with a 407
21:10.27dlynes_officesulan: if you don't force them to do sip registrations, every user can borrow someone else's username and password
21:11.06dlynes_officeand a 407 is what?
21:11.18sulan407 Proxy Authentication Required
21:11.35sulanin essence, Asterisk tries to authenticate the remote user because of the username clash
21:11.43dlynes_officesulan: and that's a bad thing?
21:12.21sulandlynes_office: well, then SIP clients with the local username foo can't call the office, even though they're registered at a foreign registrar
21:12.33SpaceBassyep, off to buy beer
21:12.33SpaceBasspeace
21:12.41Hmmhesayshave one for me SpaceBass
21:13.03dlynes_officesulan: give their remote phone a different username and password than what they're using at the office
21:13.35*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
21:13.53sulandlynes_office: well, I'm talking about any given SIP-user on the net,.. I don't think I can tell them to change the username at their registrar...
21:15.18sulan"When calling us, make sure your SIP-provider hasn't given you any of these usernames: Alice, Bob, Foo, Bar"...
21:15.41dlynes_officesulan: so what you're saying is that if someone calls into it with a given username and password, and the username's correct, but the password's not correct, and the username matches a local user, it kicks the local user out, even though the password wasn't correct?
21:16.04sulandlynes_office: no... that's not what I'm trying to say... :)
21:16.40dlynes_officesulan: can you reword it so you're a little more clear, then?
21:17.27sulanLets say we have the office pbx with a couple of SIP-phones registered. Their usernames are foo and bar, and the * server is located at office.com, so their sip-uri would be sip:foo@office.com, right?
21:17.38*** join/#asterisk tamp4x (n=Lab@64.201.13.51)
21:17.48dlynes_officeok
21:18.21dlynes_officego on
21:18.27sulanthen we have this unknown user on the net... whose username is also foo, but their registrar is located at anywhere.com, this gives sip:foo@anywhere.com
21:19.05sulanwhen he tries to place a call to sip:foo@office.com, the office.com asterisk box replies with a 407 asking the calling party to authenticate themselves, which they can't.
21:19.21*** join/#asterisk terrapen (n=cjs@166.70.183.108)
21:19.58dlynes_officeand so that user that can't authenticate themselves is not able to access your autoattendant, right?
21:20.03sulanonly because asterisk don't care about the domain name in the sip-uri, just matches any foo@<anything!> to the local user foo, which in my opinion is wrong
21:21.01sulandlynes_office: well, the invite doesn't succeed so the call can't be established - but if this unknown user changes their username to Alice or something, the call is successful
21:22.13dlynes_officetry posting something to bugs.digium.com?
21:22.23dlynes_officecheck to make sure your issue isn't already there
21:22.29sulansure...
21:23.35sulanMatching incoming calls to users and peers
21:23.36sulanAsterisk normally matches incoming calls to users based on the From: user name (without domain). However, if Asterisk can't find a user that matches the incoming call, it will try to match the caller's IP address with the IP addresses of known peers. If there's no match at all, the call will be sent to the context defined in the general section of sip.conf.
21:23.42sulanhttp://www.voip-info.org/wiki/index.php?page=Asterisk+sip+type
21:24.57sulanYou would think that it's intended when you read that paragraph... but it must be very bad, considering the scenario I have described?
21:29.05kuku5Anyone willing to debug sip with me?
21:29.26kuku5im having some nat isssues, outgoing calls dont want to come in
21:29.26sevardhahah
21:29.27sevardyeah right
21:29.43kuku5:)
21:31.12*** join/#asterisk qdk (n=qdk@x1-6-00-0f-66-90-6b-48.k441.webspeed.dk)
21:33.35[TK]D-Fenderkuku5 : Pastebin your sip.conf
21:34.13vader--hello
21:34.23vader--hows it going guys
21:38.21SpaceBasssometimes apple is stupid...i just needed to get that out
21:38.29[TK]D-FenderQwell, Damin, Hmmhesays : Confirmed presence of Polycom SoundPoint IP 430 pricing at ( www.atacomm.com ) $160 .  Not to say this is the lowest it'll go, but I'd say is still worth it.
21:38.44[TK]D-FenderSpaceBass : More like a LOT.
21:38.59*** join/#asterisk hypnox (n=dan@cornelyn.force9.co.uk)
21:39.09SpaceBassIm usually a big fan/supporter but I hate that you cannot buy songs from other counteries' music stores
21:39.24hypnoxcan anyone think why all the messages in the console might be appearing twice? It only happens after I do a reload.
21:39.30dlynes_officeSpaceBass: fruit doesn't have brains...how could an apple possibly be smart?
21:39.41SpaceBassshould have seen that coming
21:40.30dlynes_officehypnox: you've got two loggers defined in your logger.conf file?
21:40.49dlynes_officehypnox: for console messages, that is?
21:41.16dlynes_officeSpaceBass: btw...that's a thing that certain products have to protect their markets
21:41.39SpaceBassi guess...but isnt a sale a sale?
21:42.06hypnoxdlynes_office hmm as far as i can tell my logger.conf is still as the default one
21:42.13dlynes_officeSpaceBass: for instance, cars are cheaper in Canada than the US, because Canadians don't make as much money as Americans, and so a car won't yield as high a price in Canada as it will in the US
21:42.31SpaceBassso can I not buy a car in canada?
21:42.47dlynes_officeSpaceBass: sure, you can...but you have to pay all the duties to get it back across
21:42.53SpaceBassgotcha
21:43.07dlynes_officeSpaceBass: and i think there might be certain rules involved that the car has to be at least so many years old before you can bring it back
21:43.21Delta239need some hel
21:43.22dlynes_officeSpaceBass: or at least that used to be the case
21:43.25Delta239help
21:43.28Delta239on this website
21:43.30Delta239http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
21:43.38[TK]D-FenderAnd conversly Canadian weather wrecks cars so much fast than US cars and they last less....
21:43.39Delta239i am trying to do the example 2
21:43.43SpaceBasswell all I wanted was to buy some stinkin' french music!
21:44.27dlynes_officeSpaceBass: from a website in france?
21:44.32docelm0stinking is about right
21:44.35SpaceBassitunes music store
21:44.42dlynes_officeSpaceBass: in france?
21:45.03dlynes_officeSpaceBass: key word here being 'france'?
21:45.04Delta239and is not working
21:45.21SpaceBasswon't let you if your credit card billing address is outside of france
21:45.28SpaceBassand I dont have a bank account there anymore
21:45.29SpaceBassoh well
21:45.34hypnoxhmm this is wierd
21:45.39dlynes_officeSpaceBass: Yeah, anyways...it could be that way for two reasons
21:45.53dlynes_officeSpaceBass: either the people that designed the website didn't have enough forethought
21:46.19dlynes_officeSpaceBass: or because france has a lot of antiquated laws when it comes to internet and telephones
21:46.35*** join/#asterisk Renacor (n=kvirc@ip21.farheap.net)
21:46.35SpaceBassactually, its apple...and i'm sure it has to do with royalties, etc
21:46.44SpaceBassand the record companies
21:47.01dlynes_officeSpaceBass: can't you buy the french music anywhere else, then?
21:47.24SpaceBassi can find it elsewhere....i was just venting
21:47.29*** part/#asterisk _alex_mx (n=_alex_mx@200.94.154.226)
21:47.37dlynes_officeSpaceBass: like maybe itunes.ca or itunes.ag, or any other french country?
21:47.48Delta239brb
21:48.04SpaceBassapples blocks it for every country...I cannot buy music from the UK store, the Aussie store, etc
21:48.16*** join/#asterisk supjigatr (n=syslod@152.53.16.10)
21:48.31NuggetAlizeé shakes it.
21:48.46dlynes_officeSpaceBass: ah...cute
21:49.02SpaceBassalors, Alizee est choud!
21:49.18Nuggetheh
21:49.25SpaceBassdidnt think she was still around actually
21:49.27Nuggeteasy on the eyes, hard on the ears.
21:49.32Renacoris there any way to zapbarge into a channel and then send it to a phone so it rings that phone and then you can listen to that channel?
21:49.47SpaceBassanyway...Im just trying to distract myself from packing up this WIP330 and mailing it back
21:50.16dlynes_officeSpaceBass: have you tried searching on www.musiqueplus.com?
21:50.33dlynes_officeSpaceBass: or www.muchmusic.com/tv/frenchkiss/?
21:50.35*** join/#asterisk AlexCTI (n=alex@adsl-074-238-025-003.sip.mia.bellsouth.net)
21:50.36SpaceBassno, but I will, thanks
21:50.41vader--any of oyu guys doing cdr to mysql?
21:51.13dlynes_officeSpaceBass: you'll need to be able to read french if you go to musiqueplus.com though
21:51.29*** join/#asterisk nagl (n=nagl@86.59.54.237)
21:51.40dlynes_officeSpaceBass: both web sites are in canada, so there shouldn't be any cross-border problems if you're in the US
21:51.42SpaceBasspas problem
21:51.55Nuggetnah, who needs to be able to read french?  Just light up a gauloises and look disaffected.
21:51.59dlynes_officene probleme pas?
21:52.05SpaceBassLOL
21:52.38dlynes_officeWell, all the French here in Canada drink Labatt's Blue or Labatt's Blue Light
21:52.57dlynes_officeIt's because they can still ask for it when they're totally inebriated :)
21:53.05dlynes_officeboo and boo li
21:53.06*** join/#asterisk rg1_ (n=rg1@www.airlinksystems.com)
21:53.36rg1_anyone know how I can get "monitor" to play a beep for the caller to know they are recording?
21:53.55[TK]D-Fendervous etes tous le pire des francophones dans canal sti!
21:54.21Nuggetgros glandeur!
21:54.25dlynes_officetabernac!
21:54.33[TK]D-Fendersus mon pipe toi!
21:54.49[TK]D-Fender:D
21:55.05dlynes_officeunder your pipe?
21:55.17dlynes_officeerm no wait...that'd be sous, not sus :)
21:55.27SpaceBasssous
21:55.28SpaceBassyeah
21:55.38dlynes_officei don't know what the hell sus is :0
21:56.01rg1_anyone know how I can get "monitor" to play a beep for the caller to know they are recording?
21:56.23SpaceBassrg1_, does it not make the 'beep' sound now? thought that was part of monitor
21:56.41rg1_want it to beep like every 15 seconds or so
21:56.50SpaceBassahhh
22:00.47*** join/#asterisk scoody650 (n=name@h-68-165-169-170.snvacaid.covad.net)
22:01.12scoody650hello
22:01.15[TK]D-Fenderdlynes_office : So I type like shit, SHUP HOE! :D
22:02.10[TK]D-Fenderdlynes_office : Well "sus" is a diminished major dropping the 3rd  note in a major by 1 whole notes value ;)
22:02.30*** join/#asterisk sandra78 (n=aerae@200.31.115.110)
22:02.31scoody650is anyone familiar with setting up DIDs
22:03.07[TK]D-Fenderscoody650 : Yes.
22:03.56fileuh... oh...
22:04.07scoody650great.  i'm shopping out a system for a small office i will be installing asterisk in.  will the TDM040B support this?
22:04.24[TK]D-Fenderscoody650 : NO.
22:04.30scoody650what card would i need
22:04.34[TK]D-Fenderscoody650 : Analog lines do not support the concept of DID's
22:04.41*** join/#asterisk mog_work (n=mogorman@gateway.digium.com)
22:04.41scoody650i didn't think so
22:04.51[TK]D-Fenderscoody650 : And Digital trunk.
22:04.54dlynes_office[TK]D-Fender: no idea what you said...but then again, i'm not french :0
22:05.05[TK]D-Fenderdlynes_office : Which part?
22:05.28dlynes_office[TK]D-Fender: the whole diminished major shit...that sounded more like music theory than language theory
22:05.32[TK]D-Fenderdlynes_office : And FYI I just speak the language, I'm english raised.
22:05.41[TK]D-Fenderdlynes_office : I *was* talking music :)
22:05.59dlynes_office[TK]D-Fender: ah...you were talking chords :)
22:05.59scoody650[TK]D-Fender: Which card would you recommend for using DID's
22:06.06dlynes_officechoard! :)
22:07.02[TK]D-Fenderdlynes_office : indeed.  Best exemplified in "Summer of 69" by Bryan Adams in the chorus. it alternates like D, Dsus2, D, Dsus4 :)
22:07.46[TK]D-Fenderscoody650 : Any digital card.  And of Digium's TE line, or Sangoma's A1/2/4 lines
22:08.09scoody650so a DID is essentialy a digital line?
22:08.43*** join/#asterisk ManxPower (n=ewieling@stirprop-s4-0-0-21.ndcr2.datasync.net)
22:08.51*** join/#asterisk dlynes_office (n=dlynes@216.251.149.66)
22:09.06ManxPowerdoes anyone have a sound file of Allison saying "If you are sending a fax, press "start" now."
22:09.06SpaceBassa DID is a term that describes a number that can be dialed from outside your PBX that connects to your PBX
22:09.08SpaceBassmore or less
22:09.10SpaceBassin a round about way
22:09.24scoody650right
22:09.35ManxPowerDID is the same as DDI and means "telephone number direct to your extension, not requring an operator or an IVR"
22:09.39ManxPower~did
22:09.41jbotdid is, like, Direct Inward Dialing
22:09.47ManxPower~ddi
22:09.48jbotfrom memory, ddi is Direct Dialling Inward, URL: http://www.wilco-telephony.co.uk/did.html
22:10.11scoody650does it come from the telco as a RJ45 interface?
22:10.20scoody650it looks like the TE cards are RJ45
22:10.21[TK]D-Fenderscoody650 : A DID is a phone number.  these must TRAGET a digital trunk, and such calls come into the trunk with both the caller's number and the number they dailed known.
22:10.42ManxPowerThere are at least 5 ways to do DID/DDI.
22:10.49ManxPowerThe most common is using a PRI
22:11.03dlynes_officeDoes anyone know a way to get outbound calls to show up on zap show channels?
22:11.16ManxPowerdlynes_office, they already do
22:11.26dlynes_officeManxPower: on a pri?
22:11.27[TK]D-Fenderscoody650 : Before you start getting in way over your ahead and to far ahead of yourself, what do you use NOW?
22:11.28ManxPowerAhrimanes, no they don't,  they show up in "show channels"
22:11.47dlynes_officeManxPower: yeah, exactly
22:11.57*** join/#asterisk bjohnson (n=bjohnson@i216-58-51-95.cybersurf.com)
22:12.01dlynes_officeManxPower: should they not show up on zap show channels though, too?
22:12.01*** join/#asterisk quentinsf (n=quentins@cpc1-cmbg6-0-0-cust589.cmbg.cable.ntl.com)
22:12.03scoody650[TK]D-Fender: This is a new system.  AT&T reccomended a DID setup for my incoming calls
22:12.20dlynes_officeManxPower: incoming calls show up on zap show channels, but outbound don't
22:12.25scoody650[TK]D-Fender: and two outbound POTS lines
22:12.26RenacorIs there a command that you can have asterisk do a zapbarge into a channel, then call an extension to let you listen in?
22:12.32ManxPowerscoody650, I'm happy for you.  Now what type of line does AT&T recommend?
22:12.44ManxPowerRenacor, no, but you could write one
22:13.03scoody650ManxPower: I'm not sure i understand the question
22:13.13ManxPowerRecommending "a DID line" is about as useful as recommending "a Caller*ID line"
22:13.49vader--are any of you guys using mysql for configuration files instead of the flat conf files?
22:14.09*** join/#asterisk CrashHD (i=CrashHD@c-67-182-167-222.hsd1.ca.comcast.net)
22:14.20ManxPowerscoody650, if you want to use Asterisk then tell AT&T that you want "A PRI, handed to the customer as a DSX-1 interface with Caller*ID name and number".
22:14.41CrashHDanyone have a website with standard us *codes
22:14.41ManxPower..and 100 DIDs, consecutive"
22:14.45scoody650thanks, i think that makes more sense
22:14.47CrashHDlikes *69 etc...
22:14.54scoody650they quoted me two DID trunks
22:15.03ManxPowerCrashHD, you mean line the one on the Wiki, which was copied from NANPA
22:15.06dlynes_officeCrashHD: they're standard codes, period
22:15.10ManxPowerscoody650, you do not want those.
22:15.16scoody650really, alright
22:15.19dlynes_officeCrashHD: Just do a google search for NANPA vertical service codes
22:15.27CrashHDnanpa is the keyword I was looking for
22:15.27CrashHDthanks
22:15.34ManxPowersounds like they are quoting you inward only DID lines on either analog or Channelized T-1.
22:15.51scoody650i think it'll be on analog since i'm not getting a T1
22:16.01CrashHDscoody650: I've had the same problems in the past...you are dealing with sales guys...just rmemeber that
22:16.02*** join/#asterisk hinckc (n=hinckc@ool-43522ae9.dyn.optonline.net)
22:16.02ManxPowerAh.  You can't use those lines with Asterisk then
22:16.05*** join/#asterisk Qwell (n=north@unaffiliated/qwell)
22:16.14CrashHDdid trunks can not be analog
22:16.17scoody650ok, so what kind of lines then.  the PRI?
22:16.21ManxPowerdigium analog cards do not support DID
22:16.32ManxPowerCrashHD, DID trunks CAN be analog
22:16.44ManxPowerUsually delivered over E&M Wink Analog
22:16.51CrashHDnews to me
22:17.07[TK]D-Fenderscoody650 : What do you have NOW?
22:17.18dlynes_officewhat's the diff between an analog trunk and a did analog trunk?
22:17.22scoody650i have nothing but a punch block in a closet
22:17.31scoody650this is a fresh installation
22:17.35ManxPowerCrashHD, E&M Wink is an analog signaling method.  They just adapted it for use on T-1
22:17.50[TK]D-Fenderscoody650 : Ok, so you are building a PBX for a company that doesn't HAVE one right now?
22:18.03scoody650correct
22:18.05ManxPowerdlynes_office, "analog trunk" would normally be any type of analog line that does not support DID (loop start, ground start, E&M Wink, etc)
22:18.14[TK]D-Fenderscoody650 : how many users?
22:18.31dlynes_officeManxPower: so a did analog trunk is like an analog trunk, but you can deliver multiple dids over it?
22:18.32scoody6504 to start, but no more than twenty for a while
22:18.57dlynes_officeManxPower: i.e. it's effectively like a pri on a single analog line?
22:19.10ManxPowerdlynes_office, It's just a type of signaling.
22:19.16hinckcdlynes_office: always impossible to say without knowing who's saying it (each equipt mfg abuses terms differently), but the difference may be CallerID
22:19.17ManxPowernoky, nothing like a PRI
22:19.18CrashHDit has to use out of band dtmf or whatever to pass the dids right?
22:19.40ManxPowerCrashHD, there is no such thing as out of band DTMf on analog
22:19.41[TK]D-FenderManxPower : Forgot to check for you in my announcement : Polycom IP 430 potted @ $160 @ atacomm.
22:19.49*** join/#asterisk kuku5 (n=kuku5@c-71-201-217-245.hsd1.il.comcast.net)
22:20.25ManxPowerusually it's sent as something like Telco: WINK.  PBX: Wink.  TELCO; DID DTMF or DID PULSE
22:20.35scoody650[TK]D-Fender: i will only be starting with 4 users
22:20.36dlynes_officeCrashHD: isn't out-of-band a purely digital concept?
22:20.43ManxPowerI'll bet you didn't know that telcos can send the DID digits using PULSE DIALING
22:20.44CrashHDdlynes_office: ya
22:21.04[TK]D-Fenderscoody650 : Ok, Forget the concept of DID, you're talking a handful of NORMAL lines at best for that install size.  End of story.
22:21.09CrashHDdlynes_office: figured I'd difer to ManxPower. I don't deal with analog for much
22:21.11ManxPowerdlynes_office, I would have to look at the actual specs to be more accurate.
22:21.28[TK]D-Fenderscoody650 : or "plan B" of getting all of your lines provided by a VoIP provider which could offer DID's
22:21.49hinckcManxPower: yeah because the DID are the "direct inward dial" digits, and they're being delivered either by pulses or DTMF.  by why they're being delivered is either call(ing/ed) number.
22:21.54ManxPowerSome providers also do things like send DID-DTMF*CALLERID-DTMF#
22:22.09scoody650[TK]D-Fender: the client would prefer a typical POTS system.  howeer i would like the ability to expand to more numbers like a DID allows for
22:22.35[TK]D-Fenderscoody650 : For 4 its nowhere near profitable for DID capabilities unless you go pure VoIP.
22:22.38ManxPowerscoody650, If you want DID with Asterisk then you must use a T-1
22:22.50dlynes_officescoody650: you could always get them a fractional pri later on
22:22.59scoody650thanks, that nails that down
22:23.01ManxPowerOr VoIP, but only a moron would send all their calls over VoIP
22:23.02dlynes_officescoody650: when they get big enough something like that's practical
22:23.11scoody650so would i just get four phone numbers next to each other?
22:23.21[TK]D-FenderManxPower : Not without a "plan B" failover...
22:23.36ManxPower[TK]D-Fender, that is where we disagree.
22:23.46CrashHDwhy wouldn't he just go channelized t1, that's cheap and easy and he could expand and get dids
22:23.56CrashHDthats what I would be doing
22:24.04CrashHDwhy screw with telco hunt groups
22:24.05[TK]D-FenderManxPower : Truely awsome offering : manx
22:24.11scoody650compared to the quote i got from AT&T a T1 is not cost effective
22:24.12*** join/#asterisk nagl (n=nagl@86.59.54.237)
22:24.13ManxPowerCrashHD, Channelized T-1 MAY or MAY NOT be cheap.
22:24.15[TK]D-FenderManxPower : So not even with a failover hmm?
22:24.23[TK]D-FenderManxPower : http://www.polycom.com/products_services/0,1443,pw-34-182-15672,00.html
22:24.27Renacorwhere is the zapbarge command written?
22:24.30[TK]D-FenderManxPower : missed the link there.
22:24.35ManxPower[TK]D-Fender, I have my own source for Polycom
22:24.41ManxPowerHell, I'm Polycom certified.
22:24.45ManxPowerfor what that's worth.
22:24.48dlynes_officeRenacor: /usr/local/src/asterisk-1.2.7.1/apps/app_zapbarge.c
22:24.56[TK]D-FenderManxPower : About $1.50? ;)
22:24.58Renacordlynes_office: thanks
22:25.03ManxPowerdoes anyone have a sound file of Allison saying "If you are sending a fax, press "start" now."
22:25.08dlynes_office[TK]D-Fender: $1.50 for a polycom?
22:25.24ManxPower[TK]D-Fender, there's a certificate somwhere with my name on it.
22:25.25[TK]D-Fenderdlynes_office : No, the worth of ManxPower's certification ;)
22:25.27scoody650so should i purchase 4 analog lines and run them all into a TDM400P with FXo ports?
22:25.42[TK]D-Fenderscoody650 : Do you even need 4 lines?
22:25.53ManxPowerscoody650, If you were my customer I would tell you not to use Asterisk and use a dumb as rocks comdial or similar system
22:25.58scoody650with four people i think so, to allow for inbound and outbound
22:26.07dlynes_office[TK]D-Fender: it looks good when you're trying to sell polycom to a customer, even if the cert itself isn't worth anything
22:26.21dlynes_office[TK]D-Fender: the customer doesn't know that
22:26.40[TK]D-Fenderdlynes_office : I know.. and its great to be able to just grab resources yourself instead of having to ask like most other people :)
22:26.43ManxPowerdlynes_office, I don't try to sell Polycoms to a customer.  If a customer wants me to deploy VoIP then they WILL get Polycom
22:26.44dlynes_office[TK]D-Fender: same thing with MCP and MCSA and all that other crap
22:27.18scoody650if i had four lines do i jsut have them hunt to the next line if the first is busy?
22:27.25ManxPowerActually the reason is that with a certified polycom person on staff the company that sells them can get them officially rather than on the grey market and can get MUCH better prices
22:28.01[TK]D-FenderManxPower : I wasn't doubting your having your own source, jsut that their new model has hit the radar, and pics & specs are out as well as prospective pricing.  was just FYI since you love them as we do.
22:28.03dlynes_officeManxPower: and you can actually get them wholesale, then
22:28.12ManxPowerI really should do their online training to get other polycom certs
22:28.53[TK]D-Fenderscoody650 : yes, you would have them on a hunt group, and have their callerID all set to the primary #
22:28.54ManxPowerI really wish it would be legal to strangle real estate agents
22:28.55*** join/#asterisk pjchilds (n=pjchilds@pdpc/supporter/student/pjchilds)
22:28.56*** join/#asterisk dr0ck (n=dr0ck@gateway.digium.com)
22:29.05Renacordamn, can't believe there isn't a spy command that will call an extension instead of you having to call
22:29.11Nuggetdon't you mean REALTOR®s?
22:29.12*** join/#asterisk RF_MIA (n=unknown@adsl-070-147-214-250.sip.mia.bellsouth.net)
22:29.16scoody650[TK]D-Fender: ok, that makes sense
22:29.24ManxPowerNugget, Something like that.
22:29.30ManxPowerI have to deal with 350 of them
22:29.33Nuggetoof
22:29.36dlynes_officeWhy is it nobody seems to know what unicode is?
22:29.53NuggetI made a shirt.
22:29.59scoody650[TK]D-Fender: so in the event of needed axpansions i could have a series of numbers not even close, as long as theya re in the same hunt group?
22:30.35Nuggetdlynes_office: http://www.cafepress.com/nucleartacos.26746951
22:30.36dlynes_officeNugget: Yeah, but your spanish or whatever language it is you're using, is showing up as garbage characters :)
22:30.47Nuggethttp://www.cafepress.com/nucleartacos.26721820 (for mac users)
22:31.05*** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com)
22:31.06ManxPowerdlynes_office, Isn't Unicode that standard to make text take up twice as much space just so you can get letters Those Damn Foreigners Use?
22:31.31dlynes_officeManxPower: I see you're a good old boy from the sawth?
22:31.36RF_MIAunicode for R2?
22:31.58ManxPowerNugget, damn you!  I thought you made a shirt about REALTORS(r)
22:31.59[TK]D-Fenderscoody650 : sure
22:32.08ManxPowerdlynes_office, Actually I'm a yankee
22:32.15Nuggetsorry  :)
22:32.26scoody650great.  that was my main fear.
22:32.29dlynes_officeNugget: so what language was it?
22:32.34ManxPowerI was all ready to send the URL to the entire support staff
22:32.35*** join/#asterisk [hC] (n=RoadPutz@mail.rosewoodmanor.org)
22:33.00[hC]Any of you guys put two sangoma a200's in one machine ( I realize its not necessary )
22:33.10[hC]Ive configured two wanpipe configs, one for span1, and one for span2
22:33.12dlynes_officeNugget: i got REALTOR, then an A with a halo above it, then a registered trademark symbol, and then an s
22:33.26[hC]one zaptel tries to init the 5th channel (on the second card) it fails
22:33.32dlynes_office[hC]: first one is channel 1-24
22:33.43[hC]Aha.
22:33.43dlynes_office[hC]: the second card is channel 25-48
22:33.44[hC]Of course!
22:33.46NuggetLearn to use the UTF-8.  It seems to be the closest thing to a standard we'll ever get on IRC.
22:33.48[TK]D-Fender[hC] : what dlynes_office said :)
22:33.50[hC]Thank you :)
22:33.57[hC]I should have realized that.
22:33.58[hC]Btw,
22:34.00[hC]you guys may know
22:34.02dlynes_office[hC]: even if you're not using a backplane
22:34.02[hC]I was talking with sangoma
22:34.11[TK]D-Fender[hC] : Don't forget its built upon their T1 body and thats how they treat it basically....
22:34.39[TK]D-Fenderevery problem looks like a T1 to them ;)
22:34.44[hC]and they claim that if i ordered a non-echo canceller a200, i cannot simply buy an echo can module and stick it in, that i had to order the main board from them that way to begin with, because they have to program it or something. this is not what i understood.. whats the reality here?
22:34.47dlynes_office[hC]: btw...i haven't even gotten around to setting any sangoma cards up yet :)
22:35.05[hC]hah
22:35.08dlynes_office[hC]: i can let you know by Friday
22:35.11[hC]i'm putting in my first a200's
22:35.14dlynes_office[hC]: I've got one on order
22:35.18[hC]Ive got 3 of them
22:35.27[hC]but they made me order my third with echocan direct from them
22:35.30dlynes_office[hC]: no..i meant i have an a200 on order with the EC
22:35.34[hC]not just an echo can module that i snapped in to my existing cards
22:35.36[TK]D-FenderI've set up A200's before.  Nice cards (so I'm told).  I jsut did the remote install.
22:35.48[hC]they are engineered nicely, yes.
22:35.49RF_MIAdo you need echocan?
22:35.54[hC]sometimes yes sometimes no
22:36.11[hC]so you guys dont know for sure about the ecocan module requirements, etc?
22:36.14dlynes_officeRF_MIA: We got an echo can with this one, so we don't look like dumb asses should the customer get echo
22:36.32RF_MIAgood point dlynes
22:36.33dlynes_officeRF_MIA: this way it's guaranteed they don't get echo
22:36.38[TK]D-FenderGeneral question I could use a hand with : Which * package has the sample MP3's for MoH in it?
22:36.45RF_MIA"hopefully" they won't
22:36.51scoody650thanks for the help everyone
22:36.56RF_MIAI doubt its guaranteed...echo is a strange beast
22:37.00bkw__Does anyone have any more goodies on how to get a 7970G setup?  the voip-info wiki sucks
22:37.07dlynes_officeRF_MIA: yeah, but sangoma's is carrier grade
22:37.15[TK]D-FenderRF_MIA : DON'T.  Sangoma's EC is pretty rock-solid
22:37.27RF_MIAgood to know
22:37.40fileit's the Ocstasic(sp?) chip
22:37.41[TK]D-FenderRF_MIA : You'd have to be in a SICK scenario to overload that thing...
22:37.46[TK]D-Fenderfile : yup
22:37.47fileexpensive, but works well
22:38.09dlynes_officefile: yeah, but if you're popping it into a rural setting, what else are you going to use?
22:38.12[TK]D-Fenderfile : No more so than any other guys solution, a whole lot better integrated, and scales cheaper.
22:38.19fileI meant the actual chip
22:38.29filenot Sangoma product as a whole
22:38.30supjigatrThe sangoma will not have any echo.
22:38.34dlynes_officefile: it's either a digium card with a tellabs ec, or a sangoma with their ec
22:38.59rg1_In asterisk, can someone help me with the syntax of how I can Goto a "label" instead of a "priority" ?
22:39.01dlynes_officefile: both solutions are expensive
22:39.02supjigatrThe tellabs card 64ms card with daughter works well too.
22:39.28supjigatrThe a104d works great once you get it installed.
22:39.30dlynes_officebut, with sangoma's solution, you buy the ec for your first card, and it handles all the remora daughterboards, too
22:39.31supjigatr0 echo
22:40.02[TK]D-Fenderrg1_ : Goto(label)
22:40.21dlynes_officeso, it's only $230 or so for the ec, regardless of whether it's 2 lines, or 24 lines
22:40.22rg1_and how would I specify the label
22:40.26rg1_in the dialplan?
22:40.33[TK]D-Fendersupjigatr : Yup... mine's been gold from the moment I installed it.
22:40.48[TK]D-Fenderrg1_ : Go read THEBOOK.
22:40.51[TK]D-Fender~thebook
22:40.58dlynes_office~book
22:41.00jbotbook is probably a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
22:41.02supjigatrIt took me awhile to figure out that it does't play nice with slack 10.1 but once I got it on a 10.2 box it works great.
22:41.16rg1_thank you muchly
22:41.27dlynes_officesupjigatr: doesn't work on slackware 10.1?
22:41.41dlynes_officesupjigatr: or you mean with the kernel that comes with slackware 10.1?
22:41.53[TK]D-Fendersupjigatr, dlynes_office : Yeah, I'd suspect kernel
22:41.55dlynes_officesupjigatr: i.e. 2.4.37, instead of 2.4.39?
22:42.01supjigatrkernel and 10.1 udev setup
22:42.19supjigatrjust upgrading kernel doesn't work.
22:42.21dlynes_officesupjigatr: yeah...i've only ever used 2.4.39 or 2.6.15.5
22:42.44supjigatr10.2 just use test26.s and plug the card in.
22:42.46dlynes_officeerm
22:42.56dlynes_office2.4.27/2.4.29
22:43.00supjigatrI have boxes of 411p if you wanna try digium.
22:43.01dlynes_office2.4.26 was 10.0
22:43.22[TK]D-FenderSlackware 11 is due out any time....
22:43.33[TK]D-Fenderwonder when Pat will finally cave to 2.6....
22:43.38dlynes_office[TK]D-Fender yeah...stop teasing me
22:43.51dlynes_office[TK]D-Fender: it's been due out any time for about 4 months now
22:44.00supjigatrHehe. Long as test26.s is there its easy to install with 2.6
22:44.24[TK]D-Fenderdlynes_office : Thats the Slackware way!
22:44.34supjigatrHey on your 104d how are the ports labeld?
22:44.39dlynes_office[TK]D-Fender: all i know is when it's available
22:44.50dlynes_office[TK]D-Fender: i'll get it before any of you schleps :)
22:44.56[TK]D-FenderI still am scared to try and upgrade my kernel... I'm still a major linux newb...
22:45.00dlynes_officeI've got a subscription :)
22:45.33supjigatrum make bzlilo.
22:45.46dlynes_office[TK]D-Fender: change the install_path in your make file
22:45.48supjigatrI just grab the config file from slack disk and use it if the bootdisk worked.
22:46.15dlynes_office[TK]D-Fender: then do make distclean ; make menuconfig ; make bzlilo ; make modules ; make modules_install
22:46.32dlynes_office[TK]D-Fender: erm do the install_path change after the make distclean
22:46.38supjigatrand if on slack make sure it puts vmlinuz where lilo thinks it is.
22:46.50nahireananyone know why * would hang for incoming calls without RSA, but produce cause code 3 - no such context/extension for when RSA is enabled?
22:48.42ManxPowerAgents are the biggest Drama Queens
22:48.49[TK]D-Fenderdlynes_office : I may take you up on support for that later :)  However my * box IS my gateway..... so I won't be online to be guided :.
22:49.51*** join/#asterisk binhex (n=bob@216.31.167.125)
22:50.36dlynes_office[TK]D-Fender: i do remote kernel upgrades on all my machines
22:51.02dlynes_office[TK]D-Fender: occasionally i do have to do a site visit, when a particular piece of hardware is not compatible with the default kernel I install
22:55.08[TK]D-Fenderdlynes_office : I have nothing special in my box.  nvidia card (need to recompile driver, no biggie), and S518 ADSL card.
22:55.24*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
22:55.52binhexI have a ?  my * server is assigned a local and a external IP on the same card when ever my internet connection goes down all my local phones lose registration. DNS and Routing seem right. Can ne1 help?
22:57.16dlynes_office[TK]D-Fender: nvidia driver?  what for? you only do command line stuff on that box, right?
22:58.25*** join/#asterisk lesouvage (n=lesouvag@82.74.19.41)
22:59.49achandrabinhex: what does the /etc/sysconfig/network for ifcfg-eth0 say in it ?
23:00.18[TK]D-Fenderdlynes_office : Its my * server, file, FTP, web, HTPC, and used to make me coffee too :)
23:00.20lesouvageHow do I disable outbound sip calls with unregistered (soft)phones.
23:00.29achandraalso check /etc/reslov.conf for dns entry as well as /etc/host file
23:01.14binhexachandra: NETWORKING=yes
23:01.14binhexHOSTNAME=uioipbx.ioint.com
23:04.23binhexachandra: /etc/reslov.conf is enpty and /etc/host has localhost and my sip provider
23:05.23Dr-Linuxquick question, my PRI T1 provider is my US datacenter, what framing and coding i should use in zaptel.conf?
23:05.26dlynes_office[TK]D-Fender: ah...it's not that remote server you were talking about, then, with the a200 units in it
23:05.29Dr-Linuxspan=1,1,0,esf,b8zs  << is fine?
23:05.42dlynes_officeDr-Linux: ask your telco
23:05.49drrayit depends on your telco
23:05.51dlynes_officeDr-Linux: they should've told you what it was when they hooked it up
23:06.17drrayyour telco can frame it anyway you want, but you can as well
23:06.44Dr-Linuxi see
23:06.57Dr-Linuxso any signalling will be no issues with asterisk?
23:07.08drraywell, any of the big ones
23:07.21achandrabinhex: what are you using for dns resolution..dont you need resolv.conf
23:07.48Dr-Linuxactually we are already using few T1 lines from the same datacenter provider, but that's not asterisk, that's in TV
23:08.02Dr-Linuxdrray: didn't understand your last clue?
23:08.29[TK]D-Fenderdlynes_office : No, I don't care about them, this is for ME :)
23:08.36drrayI doubt a telco is going to give you a t1/pri that asterisk can't handle
23:08.39achandrabinhex: also can you let me know what the ifcfg-eth0 file has in it??
23:08.47[TK]D-Fenderdlynes_office : I have 1 linux box used a my "everything" box, and run pure VoIP here
23:09.15binhexachandra: search ioint.com
23:09.15binhexnameserver 192.168.167.3
23:09.15binhexnameserver 192.168.167.4
23:09.33binhexsorry fat figered pico :)
23:09.43dlynes_office[TK]D-Fender: so selfish :)
23:10.51achandrabinhex: two internal dns servers that looks okay
23:10.51achandrabinhex: and that ifcfg-eth0 file?
23:10.51terrapendamned extensions...for some reason, my pattern match for _011. works but _9011. does not
23:10.52terrapenit actually matches the pattern for _9011. but before I can finish dialing, the polycom sends the call
23:10.53drrayignore pat?
23:11.02terrapeneven though I have it configured not to send until the user presses send
23:11.14terrapendrray: ?
23:11.30[TK]D-Fenderterrapen : You need to make sure your polycom dialplan matches your * dialplan.
23:11.41terrapend-fender...k, i'll check that out
23:11.52[TK]D-Fenderterrapen : paste it here when in doubt
23:12.16terrapend-fender, i want the polycom to send to asterisk no matter what.  i dont want it to do its own dialplan
23:12.24terrapeni'll pastebin my dialplan for ya
23:12.42[TK]D-Fenderterrapen : So maye it appropriately generic.  tahts what I do.
23:13.11[TK]D-Fenderterrapen : #.T|X.T|*.T
23:13.37[TK]D-Fenderterrapen : thats it.  EVERYTHING requires Send/Timeout/# to terminate.
23:14.03[TK]D-Fenderterrapen : healthy that way.  I have ditched the concept of dialing 9 to dial out.  its antiquated.
23:14.50terrapenwell, our asterisk system runs in conjunction with two legacy PBXes, which use 9
23:15.06terrapenso i want to keep it standardized because my users are...uh....stuck in their ways
23:15.22achandrabinhex: any luck on that ifcfg-eth0 file?
23:15.32achandraor are you bonding interfaces?
23:15.43[TK]D-Fenderterrapen : How unfortunate...
23:15.55terrapenhttp://pastebin.com/736104
23:16.00terrapend-fender, tell me about it...
23:16.44*** join/#asterisk binhex (n=bob@216.31.167.125)
23:17.16terrapenfor some reason the _9011. matches but cuts the user off before they finish.  i set dialplan.impossibleMatch-Handling = 2 in my polycom configs
23:17.30*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
23:17.31[TK]D-Fenderterrapen [19:13] <[TK]D-Fender> terrapen : #.T|X.T|*.T
23:17.34terrapenstrangely enought, _011. matches fine
23:17.49terrapenwhat does that mean?  is that a dialplan regex of some kind?
23:18.00[TK]D-Fenderterrapen : not for *, its for your Polycom's...
23:18.05[TK]D-Fenderterrapen : to get them to STFU
23:18.08[TK]D-Fender:D
23:18.10*** join/#asterisk dlynes_office (n=dlynes@216.251.149.66)
23:18.13terrapenhrmmm
23:18.25[TK]D-Fenderist the Polycom that needs to relearn
23:18.52binhexachandra : I got booted for sending the file
23:19.36terrapenlooks like maybe I need to monkey with <digitmap/>
23:19.44terrapenbecause it has:  9]xxxxxxxxx|
23:19.52terrapenerr
23:20.00[TK]D-Fenderterrapen : Thats what I've been saying since the start.....
23:20.20terrapensorry, I was confused about what you're talking about
23:20.27terrapenlemme try that
23:20.33*** join/#asterisk RoyK (n=roy@28.80-203-106.nextgentel.com)
23:20.38achandrabinhex: use pastebin
23:21.17terrapend-fender, sorry...i do appreciate your help...i was just confused. i know nothing about polycom dialplans
23:22.21achandrabinhex: see private message
23:22.45docelm0Im about to do the unthinkable..  Im gonna install A@H!!!!!
23:23.02terrapensooooo slow
23:23.15terrapendocelm0: wimp
23:23.36docelm0terrapen if you knew who you were talking to wimp wouldnt come out of yer mouth
23:23.48docelm0Would someone clue him in on who I am please?
23:23.59sevardWho has  PRI and can I see your /etc/zaptel.conf and /etc/asterisk/zapata.conf
23:24.42[TK]D-FenderThats pre-emptive produce!  For sins to be commited!
23:24.46terrapeni thot you were serious.
23:25.12terrapenbesides, it's the internet.  you could be anybody.
23:25.29*** join/#asterisk archimedes_xyz (i=archimed@adsl-70-247-240-218.dsl.ltrkar.swbell.net)
23:25.30RoyKzoa: dang
23:25.36[TK]D-Fender:D
23:25.38*** join/#asterisk X-Gen (n=X-Gen@dsl-145-247-117.telkomadsl.co.za)
23:25.40docelm0sup tk
23:25.42gmfmsevard: i'll put mine up in a sec
23:25.49docelm0And I am serious..  Im too lazy to download and install linux
23:25.50[TK]D-Fenderheh, docelm0, ntm
23:25.54*** part/#asterisk AlexCTI (n=alex@adsl-074-238-025-003.sip.mia.bellsouth.net)
23:25.54sevardgmfm: are you in the united states?
23:25.55archimedes_xyzevening all.. anybody having problems resolving DNS for cvs.digium.com?
23:25.57gmfmyes
23:26.06sevardgmfm: what kind of setup do you have if you don't mind me asking
23:26.15docelm0And I need an asterisk box @ my house like tomorrow for my support lines..
23:26.21sevardI think I have everything figured out I just would really feel comfortable with a matching config :)
23:26.22terrapeni generally distrust people who say, "if you knew who i was..."
23:26.25terrapen:P
23:26.41[TK]D-Fenderdocelm0 : Too lazy?  That IS lame.... I do that at work (or as some would refer to it : INTEAD OF)
23:26.54docelm0nah dude..  been in the scene for 2.5 years, dCAP certified, and run one of the largest asterisk installs in the US
23:27.00terrapencool.
23:27.06[TK]D-Fenderterrapen : The statement is kinda self-defeating, isn't it :)
23:27.10terrapenwell, i'll be catching up with ya :)
23:27.17docelm0ya ya ya..
23:27.17terrapenhow many phones do you have?
23:27.18gmfmsevard: * receives PRI from Cox (actually they run fiber into our building and use an Adtran QDFR to provide PRI) and * is also providing pri_net to our Toshiba KSU
23:27.25docelm0me?
23:27.47RoyK<PROTECTED>
23:27.51terrapend-fender, yeah, you often hear the other version, "If you know who my [dad/family/lawyer] was..."
23:27.55terrapendocelm0, yeah
23:27.56docelm0ZOA!
23:28.27[TK]D-FenderCan't wait to trade up my IP 301 for an IP 430 :)
23:28.29docelm0which location?   :)   I have 90 @ my fulltime job and have an asterisk cluster of 10 dual core xeons in nyc that push well over 2M daily and I run Plainvoip
23:28.37docelm0so to count phones really doesnt mean anything
23:28.42sevardgmfm: so you have one PRI going into a TDM card? what model?
23:28.58terrapendocelm0, ok, so how big?
23:29.11terrapenok
23:29.20[TK]D-Fenderterrapen : My receptionist has that.... I'm jsut wanting the IP 430 for my HOME....
23:29.29docelm0What do you mean how big?
23:29.36terrapenwe're not that big.  I will have 400 phones in a few months
23:29.43terrapenmostly in a call center
23:29.59gmfmsevard: i'm using the varion v400p (same thing as the t400p), PRI from telco goes in, PRI to pbx goes out, plus i also use an e&m t1 to the pbx for testing
23:30.05Zodiacalis there a way to turn off the headset speaker when using dial()? when i use my overhead paging it echos whatever i say very loudly in the headset...
23:30.11docelm0Where's yer call center?
23:30.13terrapenno 2M daily or anything...but we're a retailer :P
23:30.18docelm0brb..  need to kick something
23:30.21*** join/#asterisk dlynes_office (n=dlynes@216.251.149.66)
23:30.26terrapendocelm0, salt lake city and very shortly. all over the US
23:30.35gmfmhttp://pastebin.com/736130
23:30.41sevardgmfm: so you're doing PRI -> * -> PRI, not PRI -> * -> SIP
23:30.45gmfm^^sevard
23:30.49znoGhey, does Asterisk support 484 (incomplete address response) in SIP?
23:31.18archimedes_xyzevening all.. anybody having problems resolving DNS for cvs.digium.com?
23:31.22terrapend-fender, that works like a champ, THANK YOU
23:31.27terrapenyou are the man.
23:31.45dlynes_officearchimedes_xyz: that would appear to be the case, yes
23:32.02CunningPikearchimedes_xyz: Read the list - CVS is being decommissioned
23:32.07dlynes_officearchimedes_xyz: but that is what's supposed to happen
23:32.16archimedes_xyzdlynes, thanks just wanted to make sure it's not just me.
23:32.17dlynes_officearchimedes_xyz: cvs is no more...it's been decommissioned
23:32.29gmfmsevard: yup... at this point i can't convince the owners to dump the toshiba, so i'm using * as a mediary for recording calls and logging cdrs.  I plan to start routing some outbound traffic over voip if I can ever find a carrier that is close to SoCal (voipjet is great for latency, but not so much for reliability)
23:32.34dlynes_officeoops...didn't see cp was alive there
23:32.39*** join/#asterisk achandra (n=achandra@12.44.122.130)
23:32.41sevardgmfm: facilityenable=yes ?
23:33.18sevardgmfm: also I must have missed [trunkgroups] didn't mention that in the wiki
23:33.27pjchildsznoG, looks like 484 has some mention in chan_sip.c
23:33.43gmfmsevard: hmm i think facilityenable is unnecessary now that i think about it... apparently cox did not provision our PRI initially with CID name delivery, so I tried that.
23:34.03archimedes_xyzdlynes, what list?
23:34.08dlynes_officeit's like cunningpike just comes out of the woodwork somewhere
23:34.15dlynes_officearchimedes_xyz: asterisk-announcements
23:34.29CunningPikedlynes_office: I decloak
23:34.33znoGpjchilds: nice, gonna try the early dial stuff for the Grandstream then :)
23:34.42dlynes_officearchimedes_xyz: i would imagine the asterisk homepage, too
23:34.54sevardgmfm: why don't you have usercallerid=yes and callerid=asreceived
23:34.59achandrapjchilds: thanks for your help yesterday on openser lb... the new 1.0.1 deals with failover just fine.. :)
23:35.00gmfmsevard: [trunkgroups] is necessary if you use NFAS (when you have more than one PRI from the same telco and want to share the d-channel)
23:35.02dlynes_officearchimedes_xyz: i'm really surprised you got caught though...svn's been the standard for i don't know how long
23:35.35pjchildsachandra, neat -- it would be interesting to see you ser.cfg :)
23:35.59terrapenman, i gotta get out of here
23:36.03terrapenit's mountain biking time!!!
23:37.05*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
23:37.14gmfmsevard: as it is, caller id info is passed both ways (when a call comes in, * receives caller id and sends it along to the toshiba, and when a call goes out from the toshiba, it sends the callerid of the station and * passes that to the telco)
23:37.39docelm0terra I dont necessarily define size of an intall on how many phones you have but more how many concurrent calls you push
23:37.49terrapend-fender, thanks again man.  that rocks
23:37.57terrapendocelm0, absolutely
23:38.18[TK]D-Fenderterrapen : ywc
23:38.42dlynes_office[TK]D-Fender: ywc?  your wife's cute?
23:38.45docelm0Which in my case I push on adverage 200+ channels at a slow time and upwards of 800 at peak
23:38.57docelm0s/channels/calls
23:39.07docelm0dumb bot
23:39.29dlynes_officedumb pot
23:39.32[TK]D-Fenderdocelm0 : you need to double // it like tihs
23:39.34dlynes_offices/pot/bot/
23:39.45[TK]D-Fenders/tihs/this/
23:40.15docelm0hmm didnt work in mirc
23:40.17dlynes_officedumb docelm0 :)
23:40.20[TK]D-Fenderdlynes_office : You're Welcome
23:40.24docelm0soemthing like that
23:40.27docelm0something
23:40.38Dr-Linux//
23:40.44pjchilds~spank pjchilds
23:40.45jbotACTION bends pjchilds over his knee and tatoos 'ibot' on pjchilds's pasty white buttocks.
23:41.06terrapendocelm0, we'll probably only push 3 PRI's full at a time
23:41.22terrapenmaybe more in the winter season (our big retail season)
23:41.42*** join/#asterisk mogorman (n=mogorman@68.62.237.103)
23:42.06*** join/#asterisk chino (n=Administ@c-68-84-57-212.hsd1.nj.comcast.net)
23:42.13Dr-Linuxdlynes_office: today we ordered a new DELL PE 2850 server.
23:42.19chinohi
23:42.26terrapenasterisk is currently being used as a bridge between two crappy old legacy PBXes here, which is kind of neat
23:42.28dlynes_officeoooh
23:42.41dlynes_officeDr-Linux: congratulations on figuring out how to use a web order form :)
23:42.43chinodo i need to configure dial plans if im not using fxo or fxs ports ?
23:42.45sevardgmfm: When you get a PRI from a telco is it always CPE? Or could sometimes it be pri_net ?
23:43.21[TK]D-FenderDr-Linux : So why the new server?  Growing that fast now?
23:43.40[TK]D-Fendersevard : in any sane scenario they'll play net...
23:43.43gmfmsevard: you will always be CPE to a telco, unless you come from a backwards town
23:43.49Dr-Linuxdlynes_office: thanks, but this is not our first server, we have already alot of Sun and Dell PE's
23:44.05chinohow do i tell both endpoints to communicate with one another without relaying off the server?
23:44.12sevardgmfm: are you right or is [TK]D-Fender right? :)
23:44.16dlynes_officeDr-Linux: but why is the pe 2850 such a good thing?
23:44.23Dr-Linux[TK]D-Fender: we are placing the new server at datacenter,
23:44.41docelm0Do you guys like the 2850?   Have you messed with the 1850's much?
23:44.45Dr-Linuxdlynes_office: for IVR solutions
23:44.47gmfmsevard: we effectively said the same thing... the telco is NET, you are CPE
23:45.05sevardHeh. Alright.
23:45.07znoGpjchilds: yeah, just had a look at chan_sip.c ... looks like it supports it
23:45.08Dr-Linux2850 looks good
23:45.27docelm0I bought 10 1850's the blade system they have I am very unimpressed
23:45.30sevardWell then, I think I have everything right.  I just need to wait till 7:00 A.M. when they flip the switch
23:45.35dlynes_officeDr-Linux: yeah...we always build our own servers
23:45.41gmfmsevard: when all else fails... play with it until it works :-)
23:45.44dlynes_officeDr-Linux: so i wouldn't even know what a 2850 is
23:45.47sevardYou don't need language=en on a PRI?
23:45.50dlynes_officeDr-Linux: other than that it's a rackmount
23:46.28sevardoops, yes you do :|
23:46.33gmfmsevard: there's a lot of options in zapata.conf that you don't *need*... but may affect you in certain circumstances or affect the features that you have
23:46.39Dr-Linuxdlynes_office: before ordering the server, i came here and ask for suggestion, but didn't get any answer
23:46.49sevardgmfm: ahh.  I see.
23:47.16dlynes_officeDr-Linux: yeah...nobody could probably recommend anything to you besides brand name anyways, because nobody's local to you
23:48.10gmfmsevard: one word of caution though, I had to fiddle with the pridialplan before i could make outgoing calls.  i found that unknown worked best because the telco will accept 7, 10, 11, or international digits
23:48.35Dr-Linuxdlynes_office: nope, we are purchasing in the US, not here,
23:49.05sevardgmfm: I got unknown signalling method pri_cpe
23:49.09Dr-Linuxi just explained my requirements and asked for suggestion from Sun and DELL PE
23:49.27*** join/#asterisk ManxPower (n=ewieling@stirprop-s4-0-0-21.ndcr2.datasync.net)
23:49.38dlynes_officeDr-Linux: another server that probably would have worked well is a Sun Cobalt box
23:49.39sevardgmfm:
23:49.45ManxPoweranyone having a sound file of Allison saying "If your are sending a fax, please press "start" now."?
23:49.51Dr-Linuxhhm..
23:50.10gmfmsevard: that's odd
23:50.19dlynes_officeDr-Linux: but of course they wouldn't recommend that because it runs AMD processors, not Sun processors :)
23:50.39sevardgmfm: tell me about it :\
23:50.48Dr-Linuxdlynes_office: first we were interesting in X4100 from Sun,  but that was looking low profile
23:51.14dlynes_officeDr-Linux: the cobalt boxes are their low end Linux boxes
23:51.25gmfmsevard: that came straight off the * box that i currently have processing calls... you might want to recompile and make sure everything gets built... or check google
23:51.39zpartawoohoo found a place on the net that sells voip stuff that i trust
23:51.41Dr-Linuxdlynes_office: currently we are using 2 brands DELL and Sun, for all our production servers, and PBX .. but those are not asterisk
23:51.58gmfmsevard: oh hey did you run ztcfg -vv after setting up zaptel.conf?
23:52.03sevardgmfm: I had a working TDM400P before I swapped out with this PRi card
23:52.14*** join/#asterisk binhex (n=bob@216.31.167.125)
23:52.29sevardgmfm: yeah
23:52.33loonacyI'm trying to write a PHPAGI script to display all the keys in a certain family.  Is there an AGI command to get that?  I can't seem to find one.
23:52.43Dr-Linuxdlynes_office: do you think DELL 2850 is a good choice? for 2 TE210P .. ?
23:53.04gmfmsevard: did you build/install libpri before asterisk?
23:53.09dlynes_officeDr-Linux: no idea, but any of their SunFire servers would be fine
23:53.18[TK]D-Fenderloonacy : use AMI to connect to * and issue "show database" and parse it.
23:53.30sevardgmfm: yes, like I said.. the TDM400P was workin pretty flawlessly till it was swapped out with the new PRI card :_)
23:53.40[TK]D-Fenderloonacy : Or use DB1 raw on the * DB file direct.
23:53.53dlynes_officeDr-Linux: i've only got one dell machine, and it was only cause i got a good deal on it from ebay
23:54.00gmfmsevard: libpri won't necessarily affect a TDM400P because that's not a T1/PRI card
23:54.23sevardgmfm: That's good to know.  I'll look furthure
23:55.21gmfmsevard: best of luck on the turn-up tomorrow... i get to go home now :-)
23:55.35sevardgmfm: have fun
23:55.39sevardgmfm: thanks man
23:56.42*** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon)
23:57.15Dr-Linuxdlynes_office: this server will have only IVR's in AGI.. caller will do everything via IVR.
23:57.54sevardoh crap, now where is libpri
23:58.08Dr-Linuxsevard?
23:58.14jayk-libpri?
23:58.17jayk-it should be in /usr/lib?
23:58.38sevardI don't have libpri
23:58.40Dr-Linuxyou need to compile libpri first
23:58.44jayk-you can get it from www.asterisk.org
23:59.08sevardI thought it should have came in the zaptel driver archive
23:59.26jayk-seperate
23:59.34Dr-Linuxsevard: if you are using PRI, then you need libpri, becore asterisk
23:59.37chinocan you make your voip server open or do you have to add a user section for every user in the system ?
23:59.42jayk-you are installing from package?
23:59.49chinoto the sip.conf...
23:59.51sevardDr-Linux: I understand that, where is it?

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