00:00.15 | CrashHD | anyone have any docs on regext? in sip.conf? |
00:00.21 | CrashHD | voip-info won't let me access it |
00:00.56 | *** join/#asterisk op3r (i=op3r@gr-153-202.eglobalreach.net) |
00:03.47 | redondos | Question: 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) |
00:09.24 | redondos | anyone? |
00:15.41 | *** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com) |
00:17.40 | bigmac4444 | 2 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? |
00:18.15 | robl^ | change IP addresses. sounds like maybe you have assigned the same IP to multiple devices |
00:18.24 | pjchilds | achandra, ser/openser's dispatcher appears to be stateless... so you aren't going to be able to track stuff like that... |
00:19.04 | bigmac4444 | each PC has a unique LAN IP. |
00:19.16 | bigmac4444 | both going out the same NAT |
00:19.20 | pjchilds | redondos, 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... |
00:20.13 | redondos | pjchilds: Ohh, great information. Thanks. |
00:20.53 | pjchilds | ~slap pjchilds |
00:20.55 | jbot | ACTION slaps pjchilds, keep your grubby fingers to yourself! |
00:21.35 | achandra | pjchiilds: 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.. |
00:22.21 | achandra | pjchilds: 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) |
00:23.41 | pjchilds | achandra, no idea.. we use a stateful SER with three outbound * gateways, and just try each in turn based on the response (or lack of).. |
00:24.13 | pjchilds | achandra, 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... |
00:25.11 | Flauto | what is the CUT function in asterisk? |
00:25.13 | Flauto | what does it do |
00:25.18 | Qwell[] | Flauto: it cuts strings |
00:25.30 | achandra | pjchilds: when you say stateful SER how is that configured?? |
00:25.33 | Flauto | how can it to be used? qwell |
00:25.37 | Qwell[] | show function CUT |
00:25.47 | Flauto | thank your |
00:25.57 | Flauto | whatever i want to figure out, i can just use show? |
00:26.02 | Dr-Linux | Qwell[]: Digium support was also unable to help |
00:26.20 | Qwell[] | Dr-Linux: that sucks |
00:29.02 | Dr-Linux | Qwell[] |
00:29.07 | Qwell[] | Dr-Linux |
00:29.09 | *** join/#asterisk watchy (n=watchy@h236.176.255.206.cable.cmdn.cablelynx.com) |
00:29.17 | Dr-Linux | today first time i called the Digium support :) |
00:29.32 | Dr-Linux | it was 1 hour long call , 40 minutes hold time and 20 talk time :) |
00:29.35 | InfraRed | did they tell you to FOAD? |
00:29.44 | InfraRed | :) |
00:29.53 | tzanger | nah; that wouldn't take 20 minutes |
00:29.54 | InfraRed | how much does it cost |
00:29.55 | Dr-Linux | InfraRed: FOAD? |
00:30.19 | InfraRed | Dr-Linux: http://www.urbandictionary.com/define.php?term=foad |
00:30.30 | Dr-Linux | InfraRed: cost will be, if they understand my problem. |
00:33.01 | pjchilds | Dr-Linux, what was the problem? |
00:33.05 | pjchilds | ;) |
00:33.40 | Dr-Linux | Qwell[]: they said, these cards will work on 64bits/66Mhz slots, but it's not working |
00:34.10 | Dr-Linux | he accessed my server. |
00:34.53 | Dr-Linux | pjchilds: i have 2 TE210P cards, but my system doesn't recognize them. |
00:35.19 | tzanger | you should have traded them up to a single TE410 |
00:35.40 | Dr-Linux | he used only command on my server "lspci -vv" |
00:36.20 | Dr-Linux | tzanger: nope, that's according to our business requrements. |
00:36.22 | achandra | Dr-Linux - did dmesg show anything? |
00:36.31 | tzanger | Dr-Linux: did lspci -vv show two of htem? |
00:36.39 | tzanger | actually lspci should show them |
00:36.40 | Dr-Linux | achandra: anything what? |
00:36.50 | achandra | in terms of hardware loading |
00:37.13 | pjchilds | [root@adevrg05 asterisk]# lspci -vv | grep -i Xi |
00:37.13 | pjchilds | 03:07.0 Communication controller: Xilinx Corporation Wildcard TE405P/TE410P (1st Gen) (rev 01) |
00:37.16 | Dr-Linux | tzanger: nope, it it show, then there was no need to pissed off since 2 weeks |
00:37.24 | achandra | the modules may be missing..but the hardware may show up on bootup |
00:37.42 | achandra | similar to wireless cards with no known modules... |
00:37.43 | tzanger | Dr-Linux: yeah if the cards aren't enumerated by the pCI controller no driver in the world will find it |
00:38.00 | Dr-Linux | achandra: if you even don't have asterisk installed, lspci -v should show the cards |
00:38.33 | Dr-Linux | tzanger: but alteast they must have solution |
00:38.34 | achandra | yeah...im talking at the linux level... |
00:38.48 | Dr-Linux | this is not a solution to CHANGE THE SERVER |
00:39.09 | achandra | thats what they suggested?? |
00:39.14 | achandra | lol |
00:39.19 | *** join/#asterisk FlyboySR22 (n=rsears@gateway.americanis.net) |
00:39.20 | tzanger | Dr-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 |
00:39.55 | achandra | maybe kernel doesnt detect it?? |
00:40.06 | pjchilds | is the server a HP 380 G4 ? |
00:40.06 | Dr-Linux | achandra: nope |
00:40.13 | tzanger | achandra: again this is not something hte kernel is in control of |
00:40.21 | Dr-Linux | tzanger: correct |
00:40.31 | tzanger | either the cards are enumerated when the PCI bridge is scanned or they aren't for one reason or another |
00:40.31 | Dr-Linux | well, guys |
00:40.34 | achandra | hmmm...interesting... |
00:40.56 | tzanger | sounds like a corner case that Dr-Linux got snagged on |
00:40.58 | Dr-Linux | http://206.111.151.217/dmesg |
00:41.20 | Dr-Linux | if the httpd is running, you will be able to see all system specs, i put there |
00:41.32 | pjchilds | Dr-Linux, 'connection refused...' |
00:41.53 | Dr-Linux | well, it's DELL PIII 550Mhz quad |
00:42.27 | Dr-Linux | pjchilds: yeah, just datacenter support was changing the cards's slots |
00:42.38 | Dr-Linux | so i boot the machine and didn't start the apache. |
00:42.57 | Dr-Linux | tzanger: what you suggest? |
00:43.11 | Dr-Linux | looks like i have only 2 options. |
00:43.16 | tzanger | Dr-Linux: do they show up in lspci output? |
00:43.16 | Dr-Linux | 1. change the cards |
00:43.23 | Dr-Linux | 2. change the machine. :( |
00:43.35 | Dr-Linux | tzanger: nope, never show |
00:43.51 | tzanger | use 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) |
00:45.34 | Dr-Linux | tzanger: oke wait |
00:47.32 | Dr-Linux | http://pastebin.com/734295 |
00:48.47 | Dr-Linux | tzanger |
00:49.09 | tzanger | broadcom PCI host |
00:49.11 | tzanger | that's unusual |
00:49.40 | tzanger | I wonder ... Dr-Linux in the BIOS config is there any way to lock down PCI slot speeds or otherwise tinker with it? |
00:50.07 | tzanger | that and all on one PCI bus too... that's not much of a server |
00:50.23 | tzanger | my dual xeon system has 7 PCI busses :-) |
00:50.50 | tzanger | http://pastebin.ca/58521 |
00:50.51 | Dr-Linux | tzafrir: even i called digium support and asked if i can decrease the PCI slot speeds, but he said, that's no possible :( |
00:51.01 | tzanger | Dr-Linux: look in your BIOS |
00:51.13 | tzanger | you never know what cockeyed things you can do until you poke around |
00:51.21 | tzanger | 64bit is generally at least 64MHz though :-( |
00:51.32 | Dr-Linux | :S |
00:52.15 | tzanger | however I have a TE405 in a 3.3v slot on that motherboard I pastebinned and it works |
00:52.20 | Snake-Eyes | Is there any way to stop asterisk from complaining about using inband dtmf when using g729 codec? |
00:52.20 | Dr-Linux | tzanger: i also played with my BIOS, coudn't do anything without disabling some devices, like USB, floppy, etc |
00:52.24 | tzanger | so 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) |
00:52.35 | tzanger | you can't adjust PCI timing or anything? |
00:52.57 | Dr-Linux | tzanger: no, i can't |
00:53.30 | Dr-Linux | tzanger: even i asked for to upgrade the BIOS to make that possible, but DELL support said, that's also not possible :( |
00:53.50 | tzanger | well 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 |
00:53.54 | tzanger | so I don't know |
00:54.12 | tzanger | If Digium can't make it work then they can't make it work, what more can you ask |
00:54.14 | Dr-Linux | tzanger: i think, i understand the exact problem, but just need to verify that from you. |
00:55.00 | tzanger | Dr-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. |
00:55.11 | tzanger | Dr-Linux: tell me, when you reboot the system (before linux boots) does the PCI summary list the devices? |
00:55.58 | Dr-Linux | hhm.. |
00:56.13 | Dr-Linux | tzanger: not sure about that, or not understand your question :S |
00:56.20 | tzanger | if so, rebuild your kernel to use PCI BIOS accesses instead of direct and you may be able to get by |
00:56.20 | Snake-Eyes | inband works very nicly with are setup, where rfc2833 has problems |
00:56.40 | *** join/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net) |
00:56.43 | Dr-Linux | tzanger: if cards are plugged or not, but lspci shows devices |
00:56.59 | tzanger | Dr-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? |
00:57.22 | tzanger | Dr-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) |
00:57.52 | Dr-Linux | awww |
00:58.03 | Dr-Linux | it shows only few lines, not devices |
00:58.11 | redondos | What do you think of the Linksys PAP2-NA FXS adapter? |
00:58.14 | redondos | Should I get one? |
00:58.21 | redondos | They are about 50 bucks. |
00:58.23 | tzanger | Dr-Linux: ok |
00:58.33 | syle | works fine for me |
00:58.37 | dlynes_office | redondos: they're ok |
00:58.42 | tzanger | well... recompile with PCI BIOS access only (not direct or any) and boot with the new kernel and see... |
00:58.52 | Dr-Linux | tzanger: 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 :( |
00:59.11 | tzanger | if 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 |
00:59.27 | redondos | Thanks guys. |
00:59.28 | syle | why snake-eyes? you must be using cisco gateways somewhere then |
00:59.29 | tzanger | Dr-Linux: well... shame on you for not testing before shipping the system off... not much else you can do then |
01:00.44 | Dr-Linux | tzanger: maybe we will change the machine :S |
01:00.55 | tzanger | I mean think about it though, Dr-Linux, the system's *down* right now... how much is it worth to try and get it up? |
01:01.00 | Dr-Linux | tzanger: this machine have totall 7 slots |
01:01.49 | Dr-Linux | tzanger: ok, but how can i re-biult the BIOS? if it's not possible as DELL support said |
01:02.49 | Dr-Linux | tzanger: digium will not return my cards, but they will deduct 20 % |
01:03.32 | 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 |
01:03.52 | Dr-Linux | tzanger: and right now i came to home from work, my servers are not accesible from here, PIX is there |
01:03.56 | tzanger | Dr-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 |
01:04.35 | Snake-Eyes | syle, i think are pstn termination provider does |
01:04.43 | tzanger | aha |
01:04.51 | Dr-Linux | tzanger: <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 |
01:05.05 | tzanger | Dr-Linux: try booting the linux kernel with the pci=bios kernel parameter |
01:05.17 | Dr-Linux | your 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 |
01:05.41 | Dr-Linux | yes you are right |
01:06.01 | tzanger | Dr-Linux: you can also try pci=conf1 or conf2 |
01:06.02 | dlynes_office | Dr-Linux: you could always tell your boss to hire a local tech support person for California, too :) |
01:06.06 | Snake-Eyes | syle, 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 |
01:06.07 | tzanger | it may work, it may not work |
01:06.13 | Dr-Linux | tzanger: i don't know how can i reboot linux kernel with the pci-bios kernel param... |
01:06.38 | tzanger | Dr-Linux: well you are in a spot then, aren't you... You need someone who can configure and administrate a linux box |
01:06.44 | tzanger | I think dlynes_office was offering |
01:06.51 | dlynes_office | tzanger: nope |
01:06.53 | *** join/#asterisk op3r (i=op3r@gr-153-202.eglobalreach.net) |
01:06.56 | dlynes_office | tzanger: heh |
01:07.16 | dlynes_office | I'm in British Columbia, not California :) |
01:07.27 | tzanger | dlynes_office: yeah but at least you're on the right coast. I'm in Ontario :_) |
01:07.33 | Dr-Linux | i dno't need one :) |
01:07.35 | dlynes_office | heh |
01:08.00 | Dr-Linux | till yet we have sloved all kind of issue, hope will do so, we have 50 servers over there :S |
01:08.12 | dlynes_office | Dr-Linux: well, if it helps |
01:08.38 | dlynes_office | Dr-Linux: I know someone around here that speaks urdu, and can do system administration/sql adminning |
01:09.35 | Dr-Linux | dlynes_office: dalnet is full of those type of guys :) |
01:09.35 | dlynes_office | and he has no intention of moving back to Pakistan, so you don't have to worry about job security :) |
01:09.41 | Dr-Linux | but you guys are nice. |
01:09.45 | Dr-Linux | they are not |
01:10.12 | b4ka | hey, 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 |
01:10.14 | Dr-Linux | URDU speakers never help other, |
01:10.20 | b4ka | like if i were the PSTN |
01:10.25 | dlynes_office | Dr-Linux: nah...this guy's a really nice guy |
01:10.37 | dlynes_office | Dr-Linux: i've known him since about 2 months after he arrived in Canada |
01:11.04 | Dr-Linux | dlynes_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) |
01:11.15 | b4ka | any command in asterisk to reset all the channels? |
01:11.25 | dlynes_office | b4ka: restart when convenient |
01:11.33 | Dr-Linux | if you say so.. |
01:12.01 | dlynes_office | b4ka: oh...just pri |
01:12.09 | dlynes_office | b4ka: reload chan_zap.so |
01:12.42 | b4ka | i have 2 pri lines |
01:12.50 | b4ka | i dont want to reload all of them |
01:12.55 | dlynes_office | b4ka: oh |
01:12.57 | b4ka | they t1 with the provider works |
01:13.06 | b4ka | the damnd pbx gets stuck sometimes |
01:13.15 | b4ka | we think the problem is it is waiting for the channels to reset |
01:13.25 | b4ka | it suddenly stops sending data to the asterisk |
01:13.38 | dlynes_office | b4ka: and you get busy signals dialing out on it and dialing in on it? |
01:13.52 | dlynes_office | b4ka: and you might have 2 or 3 or 4 calls that never seem to get hung up? |
01:14.10 | b4ka | kinda like that |
01:14.22 | dlynes_office | b4ka: yeah...i experienced the same problem with zaptel 1.2.5 |
01:14.30 | b4ka | oh? |
01:14.36 | b4ka | connecting to a pbx? |
01:14.45 | TripleFFFFFFFFFF | [21:11] b4ka: any command in asterisk to reset all the channels? ? |
01:14.45 | dlynes_office | b4ka: nah...just connecting to a regular pri |
01:14.47 | TripleFFFFFFFFFF | yeah |
01:14.51 | TripleFFFFFFFFFF | hit ctrl-c |
01:14.53 | TripleFFFFFFFFFF | ;) |
01:14.57 | TripleFFFFFFFFFF | stop now |
01:15.02 | TripleFFFFFFFFFF | or shutdown;) |
01:15.17 | TripleFFFFFFFFFF | soft hangup i assume you talking about ? |
01:15.22 | b4ka | well, i told you 1 pri works |
01:15.25 | dlynes_office | b4ka: I grabbed libpri-trunk and zaptel-trunk, recompiled, reinstalled, and away i went |
01:15.33 | dlynes_office | b4ka: i've never had a problem since |
01:15.36 | Dr-Linux | b4ka: your connection is stable? |
01:15.47 | b4ka | its a crossover cable of 5mts... |
01:15.49 | Dr-Linux | us soft hangup |
01:16.14 | dlynes_office | Dr-Linux: the whole port dies on him |
01:16.14 | b4ka | sometimes i can make calls |
01:16.20 | dlynes_office | Dr-Linux: erm the whole span |
01:16.21 | b4ka | then it suddenly dies |
01:16.26 | b4ka | yeah |
01:16.28 | b4ka | and only THAT span |
01:16.38 | Dr-Linux | b4ka: if the call is brigded and connection breaks, the the channels hangs like that |
01:16.39 | dlynes_office | b4ka: usually after about 22 hours or more up time, right? |
01:16.48 | b4ka | no |
01:16.53 | b4ka | like 10 minutos |
01:16.55 | b4ka | minutes |
01:16.56 | Dr-Linux | i see |
01:16.57 | dlynes_office | damn |
01:17.04 | b4ka | and when i restart sometimes it doesnt work |
01:17.13 | dlynes_office | yeah...for me it was like after three weeks the first time |
01:17.23 | dlynes_office | about 1-1/2 to 2 weeks the second time |
01:17.32 | dlynes_office | about 4 or 5 days the next time |
01:17.40 | dlynes_office | then about every 2 or 3 days for a little while |
01:17.40 | Dr-Linux | hhm.. |
01:17.50 | Dr-Linux | but i think that could be connection problem. |
01:17.51 | dlynes_office | and then every day or two |
01:18.12 | b4ka | ill pastebin some crap |
01:19.17 | Dr-Linux | b4ka: i was facing some problem. but mine was connection problem, i found later |
01:19.31 | b4ka | that happens when i make a call now |
01:19.35 | b4ka | http://pastebin.com/734339 |
01:19.50 | b4ka | Dr-Linux: its a 5mt crossover t1 cable |
01:20.07 | b4ka | and sometimes it works, so its not a bad cable |
01:20.18 | dlynes_office | yeah...looks like the same problem i was having |
01:20.26 | dlynes_office | but that's just from your log |
01:20.31 | Dr-Linux | b4ka: it's directly connection, or you have local network? |
01:20.36 | dlynes_office | your log could be caused by just about anything |
01:20.52 | b4ka | direct connection |
01:20.59 | b4ka | they are side by side |
01:21.11 | Dr-Linux | b4ka: what asterisk version? |
01:21.24 | b4ka | 1.2.5 |
01:21.39 | b4ka | errrr |
01:21.42 | b4ka | 1.2.7.1 |
01:22.15 | *** part/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net) |
01:24.16 | b4ka | asterisk-usa*CLI> pri show span 2 |
01:24.17 | b4ka | Primary D-channel: 48 |
01:24.17 | b4ka | Status: Provisioned, Down, Active |
01:24.30 | b4ka | i cant figure out why its down |
01:24.47 | dlynes_office | zap show status? |
01:24.55 | dlynes_office | do you see a yellow alarm? |
01:25.35 | Flauto | hey, dlynes |
01:25.37 | Flauto | how are you doing |
01:25.40 | dlynes_office | good |
01:25.50 | Flauto | i was reading for call record |
01:25.54 | dlynes_office | was just getting ready to head out |
01:26.00 | Flauto | anything good that i can read? |
01:26.01 | dlynes_office | reading for call record? |
01:26.05 | Flauto | yes |
01:26.08 | dlynes_office | you mean cdr? |
01:26.14 | Flauto | no |
01:26.16 | dlynes_office | or recording a conversation? |
01:26.19 | b4ka | no alarms |
01:26.24 | Flauto | yes |
01:26.28 | dlynes_office | app MixMonitor() |
01:26.40 | Flauto | how would i use it |
01:26.48 | dlynes_office | make sure you're using asterisk 1.2.7.0 or higher, too |
01:26.53 | Flauto | yes |
01:26.57 | Flauto | i have 1.2.7.1 |
01:26.58 | dlynes_office | otherwise you might find it'll segfault on you |
01:27.29 | Flauto | let me search for mixmonitor online then |
01:27.58 | dlynes_office | There's full documentation on it including examples under voip-info.org->asterisk pbx->applications->recording/playback->MixMonitor() |
01:28.53 | Dr-Linux | i'm using Monitor() app with sox package to mix |
01:29.22 | dlynes_office | Dr-Linux: mixmonitor doesn't need to spawn a separate process to mix the call legs |
01:29.35 | b4ka | dlynes_office |
01:29.38 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
01:29.48 | b4ka | what did you do to solve your problem? i dont think its the same but wth.. |
01:29.54 | b4ka | at this point ill try anything |
01:29.55 | Dr-Linux | dlynes_office: yes, but i'm using old version 1.2.0 |
01:30.34 | *** join/#asterisk zotz (n=zotz@24.231.36.9) |
01:30.37 | Flauto | okay, thanks. dlynes |
01:31.19 | *** part/#asterisk downunder33 (n=robert@219.95.251.17) |
01:31.44 | CrashHD | hey dlynes_office, can I pick your brain a bit? |
01:32.06 | dlynes_office | b4ka: i installed libpri-trunk and zaptel-trunk |
01:32.24 | dlynes_office | b4ka: they're always in a state of flux though...i just got lucky to get versions that worked properly |
01:32.55 | dlynes_office | b4ka: if you want, i can put my svn snapshots up for http for you |
01:32.59 | dlynes_office | CrashHD: what's up? |
01:33.09 | dlynes_office | CrashHD: smack |
01:33.54 | CrashHD | dlynes_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) |
01:33.55 | dlynes_office | b4ka: 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) |
01:34.19 | CrashHD | dlynes_office: same stuff we were dealing with yesterday |
01:34.55 | dlynes_office | CrashHD: You mean both the extension name and the extension number? |
01:35.16 | CrashHD | in my sip.conf I basically need to have two [221]'s |
01:35.25 | CrashHD | so that I can use those settings on different phones |
01:35.32 | CrashHD | and that text will be displayed on the phone |
01:35.42 | dlynes_office | CrashHD: answer the question :) |
01:36.04 | CrashHD | please rephrase |
01:36.12 | b4ka | what is zaptel-trunk? |
01:36.22 | dlynes_office | CrashHD: i.e. show both 221 and Reception on the phone at the same time, when it's idle, right? |
01:36.36 | dlynes_office | b4ka: it's the unstable branch of the the zaptel drivers |
01:36.44 | dlynes_office | b4ka: i.e. the development trunk |
01:36.57 | dlynes_office | b4ka: it's what will eventually become 1.2.8 and/or 1.4 |
01:37.06 | CrashHD | dlynes_office: currently all phones show the User ID that was used to register the line |
01:37.17 | dlynes_office | b4ka: erm i mean 1.2.6 and and/or 1.4 |
01:37.33 | CrashHD | dlynes_office: I need to be able to use the same user ID twice (221 x 2) |
01:37.42 | dlynes_office | CrashHD: but your user id and your caller id are the same value, right? |
01:37.45 | IceManRISK | anyone here already use jiax ? |
01:38.11 | CrashHD | on the astra ya |
01:38.25 | dlynes_office | CrashHD: try changing the caller id; see if that changes the display |
01:38.31 | CrashHD | does |
01:38.48 | dlynes_office | CrashHD: then what's the problem? |
01:38.53 | CrashHD | the display is set from the Phone number field |
01:38.57 | Sponge_bob | i 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? |
01:39.00 | CrashHD | *it doesn't I mean |
01:39.01 | CrashHD | sorry |
01:39.07 | CrashHD | hit enter mid-sentence |
01:39.30 | CrashHD | the phone number field relates to the [sip_heading] |
01:39.42 | dlynes_office | CrashHD: ok...go on |
01:39.45 | CrashHD | which seems to have to match the username=var |
01:39.55 | CrashHD | so to have a context of [221] |
01:39.56 | dlynes_office | CrashHD: are you sure? |
01:40.31 | CrashHD | dlynes_office: I registered a phone fine, changed the username=to_something_else |
01:40.35 | CrashHD | and it gave me an error |
01:40.46 | CrashHD | and tried all possibilties of combo's to use |
01:40.50 | CrashHD | to get it to reg |
01:40.59 | CrashHD | setting the auth Name to the new username value |
01:41.09 | CrashHD | and vise versa with the phone number, auth name field |
01:41.21 | dlynes_office | yeah...i don't know what to tell you then |
01:41.26 | CrashHD | heh ok |
01:41.30 | CrashHD | I picked away |
01:41.36 | CrashHD | know anyone who may be able to assist? |
01:41.42 | dlynes_office | i'm not going to have a chance to work on the aastras again for about 5 or 6 days |
01:41.47 | CrashHD | 16 hours on this one stupid thing seems retarded |
01:41.51 | dlynes_office | try Ciber311 if you see him around |
01:41.59 | dlynes_office | ~seen ciber311 |
01:42.06 | jbot | ciber311 <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'. |
01:42.09 | zwelch | CrashHD: one thought is that the phone isn't let you set your phone number/authname separately |
01:42.13 | dlynes_office | ok, guess not |
01:42.32 | *** join/#asterisk Gabriel25 (n=whyban@user-12ld5f7.cable.mindspring.com) |
01:42.33 | CrashHD | zwelch: 3 seperate phone models result in the same behavior |
01:42.34 | zwelch | i.e. that's why you have to have them be the same |
01:42.56 | dlynes_office | CrashHD: write your own firmware for them, then! |
01:43.00 | CrashHD | dlynes_office: hehe, well if you do think of anyone I would appreciate the assistance |
01:43.03 | zwelch | heh, well, i reached the same conclusion that you did; the [xx] has to equal username=xx |
01:43.18 | zwelch | but that's a problem for your scenario, i would guess |
01:43.20 | dlynes_office | CrashHD: or run asterisk on virtual hostnames :) |
01:43.27 | dlynes_office | or virtual ips |
01:43.45 | CrashHD | zwelch: but what is the additional setting on the phones for, there are three auth fields one is password and two are usernames of sorts |
01:43.47 | dlynes_office | anyways |
01:43.50 | dlynes_office | i've gotta run |
01:43.57 | dlynes_office | catch y'all later |
01:43.58 | CrashHD | thank dlynes_office |
01:43.59 | CrashHD | ttyl |
01:44.09 | Supaplex | hehe |
01:44.28 | zwelch | CrashHD: i'm not sure; i'm just speculating |
01:44.32 | CrashHD | *nods* |
01:44.39 | zwelch | but i really do have to run off to, just know that you're not alone ;) |
01:44.56 | CrashHD | so 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) |
01:55.15 | bigmac4444 | still 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) |
02:06.46 | CrashHD | how can we make the extension numbers not be displayed on the phones when the register? |
02:06.53 | CrashHD | all phones do this and it is driving me nuts |
02:07.32 | CrashHD | well the 4 I have tried today and yesterday do anyway |
02:08.01 | Gabriel25 | X-Asterisk-HangupCause: No route to destination |
02:08.12 | Gabriel25 | how I can fix this ? |
02:08.14 | CrashHD | sip or iax? |
02:08.19 | Gabriel25 | SIP |
02:08.26 | CrashHD | you have the noreinvite off? |
02:08.45 | Gabriel25 | I have no idea whats that |
02:08.57 | Gabriel25 | where I can check that |
02:08.58 | Gabriel25 | ? |
02:09.39 | CrashHD | in your sip.conf [general] |
02:09.48 | CrashHD | try doing canreinvite=no |
02:09.58 | Gabriel25 | ok |
02:10.19 | CrashHD | this will keep asterisk in the media stream (if that is the problem the error message is talking about) |
02:10.21 | CrashHD | I could be wrong |
02:11.27 | Gabriel25 | here is my sip.conf |
02:11.29 | Gabriel25 | http://pastebin.ca/58535 |
02:11.52 | CrashHD | what's in sip_nat.conf? |
02:11.56 | CrashHD | and 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-Fender | Yay more AMP! |
02:12.37 | Gabriel25 | sip_additional.conf sip.conf sip_notify.conf |
02:12.46 | CrashHD | AMP |
02:14.30 | Gabriel25 | which one CrashHD ? |
02:15.00 | CrashHD | sip_nat |
02:15.20 | CrashHD | what kind of call is this on? |
02:15.25 | CrashHD | lan to lan? |
02:15.27 | CrashHD | lan to wan? |
02:15.31 | CrashHD | wan to wan? |
02:15.35 | Flauto | how do i use mixmonitor |
02:15.43 | Qwell | Flauto: show application mixmonitor |
02:15.44 | CrashHD | ~mixmonitor |
02:17.45 | Gabriel25 | LAN |
02:18.49 | Flauto | qwell, do i need to set an extension in my dialplan or something else |
02:21.38 | [TK]D-Fender | file : You're rminding me of my favourite russian author : Imaknockoff ;) |
02:21.49 | file | hahahahaha |
02:22.33 | *** join/#asterisk fugitivo (n=ajf@201.216.246.181) |
02:22.35 | fugitivo | hi |
02:22.59 | fugitivo | i'm having serious issues with my digium cards |
02:23.09 | fugitivo | my zttest is crap |
02:23.16 | CrashHD | call digium, they love to support their equipment |
02:23.16 | fugitivo | no interrupt sharing at all |
02:23.21 | [TK]D-Fender | fugitivo : Sharing IRQ's? |
02:23.29 | fugitivo | [TK]D-Fender: no |
02:23.30 | [TK]D-Fender | fug, pastebin a dmesg. |
02:23.31 | file | incompatibility motherboard? |
02:23.47 | fugitivo | file: it's not in the list of incompatibilities |
02:24.00 | fugitivo | when i don't use a network card at all |
02:24.02 | fugitivo | it works ok |
02:24.17 | fugitivo | if I use a network card, zttest starts to fail |
02:24.19 | file | what network card? |
02:24.28 | fugitivo | e1000 |
02:24.30 | [TK]D-Fender | e100 ! |
02:24.35 | Qwell | e10000! |
02:24.42 | [TK]D-Fender | fugitivo : Thats a super-no-no for Digium cards... |
02:24.52 | fugitivo | really? |
02:25.01 | Qwell | yeah..really? |
02:25.12 | [TK]D-Fender | fugitivo : And the best part, its standard issue on a the MAJORITY of serverboards out there... |
02:25.23 | fugitivo | what a crap |
02:25.23 | file | actually, just for kicks... |
02:25.24 | CrashHD | why is it a no no? |
02:25.25 | CrashHD | technically? |
02:25.27 | [TK]D-Fender | fugitivo : Yes, I had one in mine and let me tell you.... all helll... |
02:25.41 | file | see if you can compile the E1000 driver into the compile instead of a module... |
02:25.43 | fugitivo | ok, so if i change the network card, it could work? |
02:25.43 | [TK]D-Fender | interrupt load makes Digium cards jealous ;) |
02:25.50 | fugitivo | file: i did it |
02:25.56 | file | ah, then yeah... |
02:26.02 | fugitivo | file: it works better, but fails sometimes |
02:26.11 | file | I'm a Broadcom person myself |
02:26.13 | [TK]D-Fender | fugitivo : Make sure the driver module doesn't even LOAD, and get yoursef a PCI NIC. |
02:26.17 | Qwell | realtek! |
02:26.34 | fugitivo | [TK]D-Fender: i have a pci nic and onboard nic, both e1000 :) |
02:26.56 | [TK]D-Fender | fugitivo : Good work! |
02:27.08 | CrashHD | e1000 or e100 is the problem (or both)? |
02:27.22 | [TK]D-Fender | fugitivo : And next you're going to tell me you're on an i7505 chipset MB too..... |
02:27.26 | [TK]D-Fender | e1000 |
02:27.31 | fugitivo | i'll stole a nic from the firewall |
02:27.33 | [TK]D-Fender | I typo'd |
02:28.01 | CrashHD | are 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-Fender | CrashHD : Not at all. |
02:28.44 | fugitivo | [TK]D-Fender: i think it's not, the mb wasn't on the incompatibility list |
02:28.49 | file | the 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-Fender | fugitivo : Yeah well your NIC is. |
02:29.55 | [TK]D-Fender | fugitivo : the e1000 has been a star offender for years now. |
02:30.10 | [TK]D-Fender | file : Really? What have you run into? |
02:30.28 | fugitivo | great, the firewall is using 2 3com |
02:30.32 | fugitivo | 3com is ok? |
02:30.39 | file | [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 |
02:30.41 | fugitivo | 3c905C? |
02:31.06 | file | I suppose I should be all pro-Digium butu meh, use what works! |
02:31.14 | [TK]D-Fender | fugitivo : Thats a fine card. |
02:31.20 | fugitivo | ok, i |
02:31.22 | fugitivo | i'll try that |
02:33.58 | CrashHD | where can I find out what regext= does in the sip.conf |
02:34.02 | CrashHD | voip-info won't show me |
02:34.09 | CrashHD | says I'm not that cool |
02:34.11 | [TK]D-Fender | CrashHD : its there |
02:34.21 | file | what'cha wanna know aboot it |
02:34.28 | CrashHD | just what it does |
02:34.32 | CrashHD | I know there is a option |
02:34.36 | CrashHD | that creates an extension |
02:34.38 | [TK]D-Fender | CrashHD : its damn near useless.... |
02:34.38 | CrashHD | in a context |
02:35.20 | CrashHD | it's the "damn near" part I'm worried about |
02:35.21 | CrashHD | :) |
02:35.35 | file | if you think creatively it has purposes |
02:36.23 | CrashHD | also I was hoping someone could explain what why there are three auth fields for sip devices? when only 2 seem to affect anything? |
02:36.42 | CrashHD | user, auth name, secret |
02:36.52 | CrashHD | and how those correspond to the asterisk config |
02:39.41 | znoG | never really looked into the difference between username and auth name |
02:39.41 | [TK]D-Fender | file : very little use... yeah I can picture one or two, but there are many ways to get around them without this feature... |
02:39.59 | znoG | i mean why would someone want a username different to an auth name |
02:40.03 | CrashHD | lol, you guys teasing me, or gonna tell me what it does? |
02:40.14 | znoG | i'm sympathizing damnit! |
02:40.15 | znoG | :) |
02:40.29 | CrashHD | znoG: sorry not you, file and D-Fender |
02:40.33 | CrashHD | lol |
02:40.40 | CrashHD | sympathy welcome |
02:40.43 | CrashHD | hah |
02:40.47 | file | there are reasons for the madness. |
02:41.04 | CrashHD | but those reasons aren't documented where I can find lol |
02:41.19 | file | there are also reasons for that madness |
02:41.33 | CrashHD | hah |
02:41.37 | [TK]D-Fender | CrashHD : 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. |
02:41.43 | CrashHD | small group of people trying to rule the world |
02:41.54 | [TK]D-Fender | CrashHD : clear enough? |
02:42.00 | CrashHD | perfect |
02:42.13 | [TK]D-Fender | Basically like a way to have it so that no-one can dial your exten if you aren't connected |
02:42.21 | CrashHD | fun |
02:42.23 | CrashHD | ok |
02:42.29 | [TK]D-Fender | Also known as "virtually worthless" |
02:42.34 | CrashHD | hah |
02:42.39 | file | haha... virtually.. |
02:43.12 | [TK]D-Fender | file : Yes, and like the platypus, its sole purpose is as FOOD for something higher up the chain... |
02:43.24 | [TK]D-Fender | or perhaps comic releif. |
02:43.31 | CrashHD | I vote for comic relief |
02:43.49 | *** join/#asterisk just_a_guy (n=beetle_b@74.136.209.21) |
02:43.52 | [TK]D-Fender | now that SETVAR thing in sip peers.. now THATS cool... |
02:44.03 | file | maybe, maybe not |
02:44.04 | znoG | could be a good feature though |
02:44.12 | znoG | instead of doing a ChanIsAvail, I could use regexten |
02:44.33 | [TK]D-Fender | znoG : and how would you check for it? |
02:44.45 | CrashHD | how stable is the trunk guys? |
02:44.46 | znoG | check for what? |
02:44.55 | [TK]D-Fender | znoG : Only way I know is a GOTO, and things get ugly with that line of thinking. |
02:45.00 | file | CrashHD: if you use trunk in even sort of production, I will thwap you |
02:45.01 | [TK]D-Fender | znoG : Check that it exists. |
02:45.07 | znoG | [TK]D-Fender: with chanisavail? |
02:45.11 | CrashHD | file: ok, good enough answer |
02:45.13 | file | that being said a beta for 1.4 is due to be released in... |
02:45.17 | file | 8 days |
02:45.23 | [TK]D-Fender | znoG : No, using the "regexten" method... |
02:45.25 | CrashHD | file: what about the multi parking lots for call parking? |
02:45.38 | [TK]D-Fender | znoG : ChanisAvail is a smart and FLEXIBLE command |
02:46.00 | CrashHD | znoG: better be careful...you never know who wrote what here, you may just offend someone |
02:46.03 | just_a_guy | Hi. 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 |
02:46.20 | znoG | [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? |
02:46.32 | znoG | CrashHD: huh? |
02:46.44 | Strom_C | just_a_guy: you realy shouldn't be running a desktop environment on your asterisk server |
02:46.51 | znoG | CrashHD: what did I say that could potentially offend someone? |
02:47.04 | [TK]D-Fender | znoG : 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"..... |
02:47.05 | CrashHD | znoG: Just joking, *makes a winking expression* |
02:47.28 | CrashHD | my sarcasim doesn't traverse irc very well is all |
02:47.30 | *** join/#asterisk senv (i=pjt@64.6.177.47) |
02:47.30 | just_a_guy | Strom_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. |
02:47.42 | Strom_C | asterisk@home is even worse :) |
02:48.13 | Strom_C | anyway, i wouldnt worry about NAT settings if you're running it on the same machine |
02:49.57 | znoG | [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 |
02:50.31 | just_a_guy | Strom_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-Fender | znoG : Thing is if I want to call you and you're not there.... are YOU invalid :) Thats the statement that it makes. |
02:50.57 | Strom_C | host=dynamic |
02:51.00 | Strom_C | nat=no |
02:51.20 | znoG | [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-Fender | znoG : 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! |
02:52.49 | znoG | [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-Fender | znoG : And end up in "i" because of a typo? ICK! Its a downhll trip... |
02:52.55 | just_a_guy | Strom_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? |
02:53.13 | just_a_guy | Strom_C, (And I might as well add that /var/log/asterisk/messages reports everything fine) |
02:53.18 | [TK]D-Fender | znoG : And if you INCLUDE it in another (as is the case for most IVR's) scratch that idea! |
02:53.29 | docelm0 | OI! |
02:53.29 | Strom_C | just_a_guy: x-lite is supposed to register with the asterisk box |
02:53.35 | Strom_C | so therefore, put the asterisk box's IP |
02:53.50 | [TK]D-Fender | znoG : 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) |
02:54.03 | just_a_guy | Strom_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) |
02:55.10 | znoG | [TK]D-Fender: can't I do VoicemailMain(${EXTEN}) ? actually ${EXTEN} == i wouldn't it... |
02:55.11 | Strom_C | just_a_guy: or maybe your sip.conf is munged up |
02:55.20 | [TK]D-Fender | znoG : BINGO! |
02:55.22 | *** join/#asterisk fugitivo (n=ajf@201.216.246.181) |
02:55.23 | znoG | [TK]D-Fender: i'm now convinced, i shall stick to ChanIsAvail |
02:55.30 | fugitivo | ok |
02:55.39 | znoG | [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-Fender | znoG : Congrats! Hard learned knowledge! |
02:56.07 | [TK]D-Fender | znoG : I thought it was cool too... then I blinked and saw the light :D |
02:56.26 | znoG | [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 |
02:57.00 | CrashHD | voip-info should have a version tag for what asterisk version was current when something was posted |
02:57.01 | znoG | [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-Fender | znoG : thats could be useful I guess... certainly more than regexten... |
02:58.47 | just_a_guy | Strom_C, Could be, but my sip.conf is bare minimum... |
02:59.00 | fugitivo | [TK]D-Fender: i changed the card |
02:59.06 | fugitivo | [TK]D-Fender: and same problem |
02:59.26 | [TK]D-Fender | fugitivo : you need to completely disable the e1000 AND remove the kernel module |
02:59.48 | [TK]D-Fender | fugitivo : its not enough to leave it disconnected from the lan |
02:59.55 | fugitivo | i did |
02:59.56 | fugitivo | wait |
03:00.02 | fugitivo | now i have an irq sharing |
03:00.06 | fugitivo | damn |
03:00.12 | fugitivo | i can't finish with this server :) |
03:00.21 | Strom_C | just_a_guy: pastebin the sip.cong |
03:00.22 | Strom_C | er |
03:00.26 | Strom_C | sip.conf |
03:00.28 | Strom_C | ~pb |
03:00.30 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
03:02.11 | just_a_guy | Strom_C, [general] |
03:02.11 | just_a_guy | context=default ; Default context for incoming calls |
03:02.11 | just_a_guy | bindport=5080 ; UDP Port to bind to (SIP standard port is 5060) |
03:02.11 | just_a_guy | bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) |
03:02.11 | just_a_guy | srvlookup=yes ; Enable DNS SRV lookups on outbound calls |
03:02.12 | just_a_guy | [myxtenid] |
03:02.14 | just_a_guy | type=friend |
03:02.16 | just_a_guy | secret=welcome |
03:02.18 | just_a_guy | qualify=yes |
03:02.20 | just_a_guy | nat=no |
03:02.22 | just_a_guy | host=dynamic |
03:02.24 | just_a_guy | canreinvite=no |
03:02.25 | CrashHD | pastebin is your friend |
03:02.26 | just_a_guy | context=default |
03:02.33 | CrashHD | http://www.pastebin.com/ |
03:03.29 | Strom_C | just_a_guy: I said pastebin |
03:03.43 | Strom_C | ~pb |
03:03.44 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
03:04.48 | just_a_guy | Strom_C, Apologies, didn't know what pastebin was: http://pastebin.com/734442 |
03:05.20 | CrashHD | just_a_guy: no worries mate, gotta start somewhere |
03:05.38 | fugitivo | is 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-Fender | fugitivo : Sure, why not.... |
03:06.50 | fugitivo | ok |
03:09.05 | *** join/#asterisk JunK-Y (n=junky@modemcable205.175-81-70.mc.videotron.ca) |
03:09.54 | senv | i am haing wierd problems with my tdm400p. incoming lines connect then hangup |
03:10.12 | senv | is this a common problem? |
03:10.47 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-51-95.cybersurf.com) |
03:10.48 | senv | keeps getting zap at incoming failed |
03:11.12 | senv | i can dial out on the lines fine |
03:12.12 | Strom_C | senv: what context do the fxo channels live in, and do you have an s extension in that context in nyour extensions.conf? |
03:12.21 | fugitivo | senv: 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-Fender | senv : pastebin the call attempt at CLI on verbose 10 |
03:13.26 | [TK]D-Fender | ~pb |
03:13.27 | jbot | i 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) |
03:16.16 | senv | ok it is http://pastebin.com/734454 |
03:17.32 | Gabriel25 | can someone can help me |
03:17.44 | fugitivo | senv: pastebin your extensions.conf ONLY if your |
03:17.50 | Gabriel25 | I what somene to try to connect from an external ip to my asterisk box |
03:17.52 | fugitivo | you're not using asterisk@home |
03:17.59 | fugitivo | ^^^ senv |
03:18.16 | senv | i am using asterisk@home |
03:18.25 | fugitivo | lol |
03:18.35 | senv | freepbx |
03:18.43 | Strom_C | go to #freepbx |
03:18.48 | senv | lol. o k |
03:18.51 | senv | thanks :) |
03:19.06 | *** part/#asterisk senv (i=pjt@64.6.177.47) |
03:19.44 | fugitivo | [TK]D-Fender: still the same problem :( |
03:22.04 | [TK]D-Fender | fugitivo : Did you disable the E1000 in the bios? |
03:22.41 | [TK]D-Fender | fugitivo : 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) |
03:22.53 | hacked`` | guys |
03:23.01 | hacked`` | 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 |
03:23.06 | hacked`` | and it would route the call to the right person |
03:23.08 | hacked`` | what do i need for that |
03:23.17 | fugitivo | [TK]D-Fender: disabled from the bios and from the kernel |
03:23.18 | znoG | Asterisk !! |
03:23.24 | fugitivo | [TK]D-Fender: and rebooted |
03:23.28 | hacked`` | znog, ya but i mean in terms of hardware |
03:23.43 | znoG | well that depends on many things |
03:23.57 | InfraRed | hacked``: depends on how you want to connect to the pstn |
03:24.01 | fugitivo | [TK]D-Fender: i'm using a 3com right now |
03:24.01 | znoG | like 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 :) |
03:24.09 | hacked`` | infrared, what are my choices here |
03:24.18 | InfraRed | we're not sales people |
03:24.19 | hacked`` | znog, correct |
03:24.25 | InfraRed | do your onw research |
03:24.27 | InfraRed | own |
03:24.31 | hacked`` | infrared, this is part of my own research |
03:24.42 | InfraRed | voip-info.org then |
03:24.47 | znoG | i'm willing to bet this was your first option in your "research" :) |
03:25.03 | hacked`` | actually i read the asterisk wiki, and its all gibberish to me |
03:25.03 | InfraRed | reasearch by asking someone else |
03:25.09 | znoG | if 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-Fender | fugitivo : Hmmmmm... |
03:25.24 | znoG | it's easy to say "i want X and Y to do Z, what do I do?" |
03:25.34 | hacked`` | im not asking you how to set up asterisk, im asking what i need, in general |
03:25.55 | [TK]D-Fender | hacked`` : Depends. What kind of lines are you intending on using? |
03:26.41 | InfraRed | he plans to spam people with phonecalls! |
03:26.42 | fugitivo | [TK]D-Fender: a big hmmmmmmmmm |
03:26.46 | hacked`` | 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 |
03:27.04 | InfraRed | how is that incoming line presented? |
03:27.13 | InfraRed | analogue/isdn |
03:27.15 | ManxPower | These business trips really put the miles on the car |
03:27.20 | hacked`` | its just a regular analog line |
03:27.34 | InfraRed | get a digium fxo card then |
03:27.48 | hacked`` | what about an fxs card |
03:27.54 | InfraRed | a server and couple of ip phones |
03:28.00 | InfraRed | what about it |
03:28.02 | InfraRed | they're nice |
03:28.07 | hacked`` | why dont i need one of those |
03:28.12 | InfraRed | rtfm its all described there |
03:28.15 | hacked`` | doesnt make sense that i would need 1 fxo but not a fxs |
03:28.20 | ManxPower | Well, a Digium TDM400P w/FXO or a SIPura SPA-3000, or a Clone X100P card, or, or, or |
03:28.22 | InfraRed | r t f m |
03:28.33 | InfraRed | esp the fxo vs fxs part |
03:28.38 | hacked`` | k |
03:28.39 | ManxPower | ~fxofxs |
03:28.40 | jbot | it 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. |
03:28.42 | InfraRed | voip-info |
03:28.45 | [TK]D-Fender | hacked`` : any of what ManxPower jsut suggested + the A200. |
03:29.02 | ManxPower | [TK]D-Fender: heritic |
03:29.12 | InfraRed | A200 ? |
03:29.14 | [TK]D-Fender | ManxPower: zealot |
03:29.20 | [TK]D-Fender | :D |
03:29.26 | ManxPower | [TK]D-Fender: 8-) |
03:29.26 | InfraRed | whats an A200 |
03:29.29 | InfraRed | ~A200 |
03:29.38 | ManxPower | [TK]D-Fender, I'm planning on going with Sangoma for my next project |
03:29.53 | [TK]D-Fender | ManxPower : hehe, fine! |
03:30.00 | [TK]D-Fender | ManxPower: hypocrit |
03:30.00 | [TK]D-Fender | :D |
03:30.01 | ManxPower | InfraRed, Sangoma. They make cards that work with Asterisk using their own drivers to emulate the Digium interface. |
03:30.07 | InfraRed | ah nice |
03:30.14 | ManxPower | [TK]D-Fender: Kids, don't do drugs! |
03:30.31 | InfraRed | cisco phones are overrated |
03:30.34 | hacked`` | 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-Fender | ManxPower : Stay in milk! Don't do school! Drink your drugs! |
03:30.41 | InfraRed | they;re PITA to install, PITA to configure |
03:30.56 | [TK]D-Fender | InfraRed : Not so must overrated as overpriced. |
03:31.11 | *** join/#asterisk voipaster (n=25x8supp@203.167.120.9) |
03:31.20 | [TK]D-Fender | hacked`` : You can get the ringtone working on ALL SORTS of phones... |
03:31.25 | ManxPower | hacked``, Get Polycom. |
03:31.26 | fugitivo | i think i'm not going to buy digium cards anymore |
03:31.36 | voipaster | hi |
03:31.42 | [TK]D-Fender | hacked`` : Linksys SPA9xx have it as an option by default as they are made by Cisco. |
03:31.44 | InfraRed | hacked``: cisco is more hassle than its worth |
03:31.48 | ManxPower | Ciscos don't come with SIP firmware (extra cost) and don't come with a power supply (extra costs) |
03:31.55 | [TK]D-Fender | hacked`` : Yup, a huge thumbs up for Polycom. |
03:32.04 | voipaster | im new on asterisk, im installing my digium card te110p |
03:32.32 | voipaster | can someone give me a step by step on installing and configuring it pls |
03:32.55 | InfraRed | voipaster: www.voip-info.org |
03:32.57 | voipaster | ok thanks |
03:32.58 | InfraRed | ~docs |
03:33.00 | jbot | methinks 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 |
03:33.05 | InfraRed | its all there |
03:33.07 | InfraRed | ^^^ |
03:33.21 | ManxPower | ~thebook |
03:33.28 | InfraRed | dont expect this to take few hours |
03:33.31 | InfraRed | plan days |
03:34.02 | ManxPower | It was hard for us to learn, it should be hard for you to learn. |
03:34.45 | Strom_C | hire a consultant to do it for you |
03:34.53 | Strom_C | I'm a bargain at only $125 per hour |
03:36.04 | [TK]D-Fender | Was easy for me.. but I'm an adept :) |
03:36.06 | ManxPower | I'll do it for $120/hr as long as I can insult you too. |
03:36.22 | Qwell | I'll do it for $115/hr as long as I can insult ManxPower too. |
03:36.34 | JunK-Y | mouhahah |
03:36.35 | ManxPower | "Here's my bill. You're ugly and your mother dresses you funny." |
03:36.47 | [TK]D-Fender | Qwell : Heck, I do that for FREE! Am I not truely AWESOME? |
03:37.37 | Strom_C | For $125 per hour I throw in insults to the idiot who set up the system that I'm rebuilding and setting up correctly :) |
03:37.55 | Qwell | Strom_C: you win |
03:38.02 | InfraRed | cheap bastards |
03:38.20 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
03:39.29 | [TK]D-Fender | voipaster : Before even talking about your TE110P, how much have you worked with *? |
03:40.02 | InfraRed | [TK]D-Fender: he just said he was new |
03:40.16 | gandhijee | whoever was thinkin about polycoms, don't get them if u want to use the microbrowser to actually do shit |
03:40.48 | [TK]D-Fender | gandhijee : Have you talked to their techs about it? |
03:40.49 | ManxPower | gandhijee, only the 600 has a microbrowser anyway. |
03:41.01 | gandhijee | yah i know |
03:41.07 | ManxPower | if you want a microbrowser then go with cisco and pay the extra money |
03:41.28 | gandhijee | [TK]D-Fender: i emailed them yesterday from that knowledge page they had, tryin to get some documentation on it |
03:41.59 | gandhijee | we'll see what happens |
03:42.36 | [TK]D-Fender | gandhijee : well the double-post issue should be traceable |
03:42.50 | gandhijee | i really don't think thats in my code |
03:43.00 | CrashHD | anyone have an updated website about running multiple asterisk installations |
03:43.15 | gandhijee | i think it has to do something with the way the polys handle forms |
03:43.16 | [TK]D-Fender | gandhijee : I didn't think so either.... so they shoudl be able to figure it out. |
03:43.18 | Gabriel25 | guys I have a problem .... when I try too add an external SIP phone dosen`t work |
03:43.27 | [TK]D-Fender | gandhijee : did you try to break it down like I suggested? |
03:43.29 | Gabriel25 | what can be the problem ? |
03:43.31 | gandhijee | yeah |
03:43.48 | ManxPower | Gabriel25, it could be a billion differnet things. |
03:43.52 | gandhijee | for the XML error thing, i think the page was just too large |
03:44.07 | gandhijee | too much code for it to handle, which seems kind of retarded |
03:44.13 | [TK]D-Fender | ManxPower : Especially since he's using AMP.... |
03:44.17 | ManxPower | assuming Asterisk is on a public IP address, all you should need is nat=yes |
03:44.20 | Gabriel25 | ok my linux box have 2 network cards one is LAN and one WLAN from LAN is working fine from WLAN dosen`t work |
03:44.23 | gandhijee | but then again i should be breakin some of the stuff down in to modules/functions |
03:44.26 | hacked`` | why should it be hard |
03:44.33 | ManxPower | [TK]D-Fender, Ah, then it could be a quadrillion different things |
03:44.59 | [TK]D-Fender | ManxPower : I believe the appropriate term is "arbitrarily large number" ;) |
03:45.02 | Gabriel25 | ManxPower where I can put nat=yes? |
03:45.13 | ManxPower | Gabriel25, Is the phone behind NAT? |
03:45.18 | Gabriel25 | in the server config files or on the SIp soft phone ? |
03:45.32 | Gabriel25 | I don`t know how to do this |
03:45.36 | ManxPower | Gabriel25, all NAT for clients is in sip.conf. |
03:45.41 | ManxPower | Gabriel25, read The Book |
03:45.49 | gandhijee | Gabriel25: so your softphone doesn't work in the WLAN? |
03:45.51 | ManxPower | ~thebook |
03:46.01 | jake1932 | jbot is dead |
03:46.17 | pjchilds | ~slap jbot |
03:46.18 | jbot | ACTION slaps jbot, keep your grubby fingers to yourself! |
03:46.54 | jake1932 | or not |
03:47.51 | Gabriel25 | Thank you ManxPower |
03:48.27 | ManxPower | If you set nat in asterisk and nat in the SIP client, well two nats don't make it right. |
03:49.36 | pjchilds | anyone use a session-border-controller gateway in-the-real-world ? |
03:50.46 | docelm0 | yes |
03:50.47 | docelm0 | why? |
03:54.58 | pjchilds | docelm0, just what type of SBCs people were using, and what they thought of them... |
03:54.58 | Gabriel25 | ManxPower thank you is working now I had to change from nat=never |
03:55.09 | Gabriel25 | to nat=yes |
03:55.12 | Gabriel25 | and is working ! |
03:55.15 | docelm0 | MERA can suck my D!CK.. I HATE IT! |
03:55.22 | Gabriel25 | thank you so so much |
03:55.34 | docelm0 | its one of the worst.. but MVTS II is supposed to be better |
03:56.39 | docelm0 | actually the worst would be PortaOne.. They suck big time.. they call themselves a SBC but they are not even close |
03:57.20 | docelm0 | I messed with nextone.. it wasnt too bad.. |
03:57.38 | pjchilds | amcepacket? |
04:00.18 | [TK]D-Fender | pjchilds : I wouldn't trust a company starting with acme.... I remember this one poor coyote... |
04:00.28 | Qwell | [TK]D-Fender: That was a different acme |
04:00.52 | jake1932 | it was ACME not acme |
04:01.12 | docelm0 | The only guys who use acme I know of are GX and XO |
04:01.31 | fugitivo | this is a mess |
04:01.41 | fugitivo | 99.902344% 99.902344% 99.890137% 99.743652% 99.536133% 99.902344% 99.902344% 99.890137% |
04:02.42 | pjchilds | fugitivo, time to get a sagnoma ? :) |
04:03.14 | znoG | fugitivo: i get 99.70% on a FXO card... one fax out of 10 make it |
04:03.26 | znoG | i have to make it go on IRQ 9 which is not the easiest thing to do |
04:03.29 | znoG | on this mobo anyway |
04:03.30 | fugitivo | time to escape to another country |
04:07.58 | fugitivo | digium support is great |
04:08.19 | fugitivo | what is the response time for emails? 1 week? |
04:08.23 | CrashHD | zttest should be 100% all the time right? |
04:08.38 | fugitivo | it should |
04:08.44 | znoG | well i get 99.95% or something which I consider quite cood |
04:08.45 | znoG | good |
04:08.49 | fugitivo | between 99.98 and 100 |
04:08.49 | JunK-Y | CrashHD: not necesseraly |
04:08.51 | znoG | occasionally it hits 100% but rarely |
04:09.08 | CrashHD | ahh ok |
04:09.40 | CrashHD | I have an e1000 in that box and it is still 99.8+ |
04:15.36 | gandhijee | after i loaded irqbalance i got better results on my box |
04:15.45 | asterboy | Anyone setup a Bogen Lucent paging system? |
04:15.49 | gandhijee | but its a hodge podge of hardware |
04:15.49 | *** join/#asterisk meesterfox (n=M_fox@71.224.224.168) |
04:16.03 | asterboy | Playing with a LUPCMALL |
04:16.32 | asterboy | Has TIM,CPU,TBM and ZPM modules |
04:16.34 | meesterfox | Anyone have any experience with the asterisk wakeup call annoy script? or one of it's variants... |
04:17.04 | asterboy | Just want to know if there is a way to hookup a speaker without an AMP. |
04:17.22 | asterboy | Manual says you can use 70V |
04:17.52 | gandhijee | hey can someone take a look at this error |
04:17.52 | gandhijee | http://pastebin.com/734494 |
04:18.03 | asterboy | Not sure if I should use an FXO or FXS module to connect it. |
04:18.08 | asterboy | wants loopstart |
04:18.10 | gandhijee | its giving me some crap about ODBC and bad SQL |
04:18.19 | asterboy | so I'm thinking FXO |
04:19.26 | asterboy | otherwise, I'll just hookup something to the sound port of the * box. |
04:19.34 | asterboy | problably 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) |
04:28.47 | supjigatr | Hi. |
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) |
04:31.39 | sudhir492 | Hi all |
04:31.54 | sudhir492 | anyone using spandsp and rxfax heres? |
04:33.26 | supjigatr | Yep |
04:33.31 | *** join/#asterisk sudhir492 (n=sudhir@pool-71-114-99-2.washdc.dsl-w.verizon.net) |
04:33.42 | supjigatr | Well I was till my sangoma driver stop loading after a reboot. |
04:33.47 | sudhir492 | Awfully quiet in here |
04:33.49 | mitcheloc | isn't spandsp replaced by asterfax now? |
04:33.50 | supjigatr | Yep |
04:34.01 | supjigatr | huh? |
04:34.01 | sudhir492 | asterfax, hmm |
04:34.07 | supjigatr | spandsp is asterfax |
04:34.14 | sudhir492 | let me check. I thought that asterfax uses spandsp |
04:34.21 | supjigatr | It 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) |
04:35.02 | the_real_JasonF | Hello all |
04:35.06 | supjigatr | Hi |
04:35.08 | fugitivo | what is asterfax? |
04:35.17 | mitcheloc | try google? |
04:35.33 | the_real_JasonF | I am trying to redirect incoming calls to an outside number, but unless I pick up the call it is failing |
04:35.44 | supjigatr | Before I kick this machine, does anyone have a sangoma card a104d working? |
04:35.46 | the_real_JasonF | is there anyway to answer and call and redirect it automatically...? |
04:35.52 | supjigatr | Yea |
04:36.02 | supjigatr | dial |
04:36.17 | the_real_JasonF | dial doesn't pick up the call.. |
04:36.26 | [TK]D-Fender | supjigatr : I do, what of it? |
04:36.43 | sudhir492 | yes |
04:36.54 | sudhir492 | I have sangoma A104 working |
04:36.56 | the_real_JasonF | maybe I should just Answer, then dial with r |
04:36.59 | sudhir492 | works great |
04:38.13 | the_real_JasonF | hehe, answered my own question.. |
04:38.27 | the_real_JasonF | but I wonder why I need to answer :-( |
04:38.49 | supjigatr | I can't seem to keep it working. |
04:39.00 | sudhir492 | the_real_JasonF: Try something like |
04:39.00 | sudhir492 | exten => 7034441234,1,Dial(Zap/g1/12124567890,25) |
04:39.35 | meesterfox | anyone have any experience with the wake up call script? |
04:40.31 | meesterfox | mine works fine, except when you hangup it doesn't call you back or anything. |
04:40.32 | supjigatr | D-Fender: Are u using the ec and hdlc on the card? |
04:41.41 | supjigatr | I suspect its UDEV problem but I can't seem to figure it out. |
04:41.44 | sudhir492 | Unfortunately, I do not remember what I configured the card with. But it works great so far |
04:41.59 | sudhir492 | just follow Sangoma's instructions |
04:42.20 | supjigatr | I followed them. I'm running slackware. |
04:42.32 | supjigatr | Are you using UDEV? |
04:42.37 | sudhir492 | hmm. I am running FC3 |
04:42.38 | supjigatr | linux 2.6.x? |
04:42.48 | [TK]D-Fender | supjigatr : you, the works. |
04:43.35 | supjigatr | I want it all too but I had it working once but after reboot it fails. |
04:44.42 | the_real_JasonF | <supjigatr> < I actually tried just a dial, but it gets rejected |
04:44.43 | supjigatr | Right now I get this wanpipe FATAL: Error inserting af_wanpipe |
04:45.23 | the_real_JasonF | the only difference I can see is that the IP from IP changes to the asterisk server ( or in internal IP if I redirect manually) |
04:45.51 | the_real_JasonF | but if i Answer it in the dialplan, I figure it means the callers starts getting charged.. |
04:46.32 | the_real_JasonF | ie. dials out via sip, no POTS |
04:46.37 | sudhir492 | yes, Linux 2.6 |
04:49.06 | [TK]D-Fender | supjigatr : where do you get that error? |
04:49.52 | supjigatr | wanrouter start |
04:50.45 | *** join/#asterisk jeebusroxors (n=jeebusro@29palms-cuda1-68-170-36-65.losaca.adelphia.net) |
04:50.50 | [TK]D-Fender | sounds like amke a permissions thing... |
04:50.56 | [TK]D-Fender | maybe* |
04:51.09 | supjigatr | Where? |
04:52.01 | *** join/#asterisk Crshman (n=chatzill@hacienda-heights-cuda2-68-71-5-62.lmdaca.adelphia.net) |
04:52.02 | [TK]D-Fender | supjigatr : not entirely sure.. |
04:52.09 | Crshman | if 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-Fender | Crshman : What is this magical "follow me" of which you speak? |
04:53.29 | Crshman | ooooo that's right it's only in FreePBX, oops |
04:53.38 | InfraRed | stalker mode pbx |
04:53.39 | Crshman | ok wrong channel then sorry |
04:53.51 | *** join/#asterisk angler- (n=angler@pdpc/sponsor/digium/angler) |
04:54.22 | [TK]D-Fender | Crshman : No, only the version YOU are talking about... |
04:54.29 | *** join/#asterisk iceyp (n=icepick@firewall.unix.co.nz) |
04:54.39 | Crshman | [TK]D-Fender: ? i don't understand |
04:54.54 | iceyp | hey 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-Fender | Ok, I'm fried.. back tomorrow peeps.... |
04:56.26 | iceyp | should 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? |
04:56.27 | JunK-Y | iceyp: show g729 |
04:56.51 | *** join/#asterisk jero (n=jero@modemcable235.87-82-70.mc.videotron.ca) |
04:57.09 | iceyp | the 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 |
04:57.37 | Strom_C | iceyp: one license to talk to the phone, one license to talk to the outside world, and then oops, out of licenses! |
04:57.56 | iceyp | stoffell then it should go to ulaw or something? |
04:58.14 | iceyp | it uses 2 licenses even when i call voicemail or meetme |
04:58.23 | Strom_C | use ulaw locally |
04:58.35 | iceyp | my pabx is remote to me |
04:58.45 | iceyp | so i use 729 to the pabx |
04:59.03 | iceyp | but if there is no codecs it should drop to ulaw as next option, rather than not let me make a call at all |
04:59.16 | SwK | you dont need 2 licenses of G729 to talk thru the PBX |
04:59.27 | jero | anyone experienced issues with voip without qos on a 100mbps lan ? |
04:59.28 | SwK | you only need 1 license for each transcoding session |
04:59.40 | SwK | jero: only when the LAN is heavily loaded |
05:00.07 | jero | swk: what is heavily? 1 transfer between 2 hosts ? |
05:00.16 | jero | or many many ones |
05:00.48 | SwK | 1 transfer between 2 hosts can congest the network... it really depends on what type of hub/switch you have and what the hosts are |
05:01.20 | SwK | then again some networks take many transfers between many hosts to get truely congested |
05:01.33 | SwK | it all depends on the hosts and related network hardware |
05:02.05 | iceyp | stupid stupid codecs |
05:02.21 | Strom_C | g729 blows anyway |
05:02.30 | litage | can a linux box's hostname begin with an underscore? |
05:02.51 | Qwell | litage: I don't see why not |
05:02.57 | SwK | i'm sure it could start with whatever printable ascii you wanted it to |
05:02.59 | iceyp | if i have no codecs specified in sip.conf [general] section, will this stop all calls? |
05:03.08 | jero | thanks swk |
05:03.12 | Snake-Eyes | Is 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. |
05:03.26 | Strom_C | Snake-Eyes: you cant use inband dtmf on g729 |
05:03.38 | Strom_C | at least not with any hope of reliability |
05:03.54 | litage | thanks Qwell |
05:03.54 | SwK | DTMF inband with g729 might work but its far from usable in most situations |
05:03.55 | Strom_C | the codec fucks up the sine waves |
05:04.17 | Snake-Eyes | well i am and seems work fine so far. |
05:04.29 | SwK | what rfc2833 issue are you having? |
05:04.50 | Strom_C | Snake-Eyes: if you want inband dtmf, use ulaw |
05:04.56 | SwK | or alaw |
05:04.59 | Strom_C | otherwise, use info or rfc2833 |
05:05.00 | Snake-Eyes | is it posiable that asterisk is complaining that the one channel is g729, but channel going out isnt |
05:05.36 | iceyp | doubt it |
05:05.43 | Snake-Eyes | eg channel to end pt is ulaw and channel coming back is g729 ? |
05:05.58 | iceyp | seems that asterisk is using the codecs inlisted in my [general] rather than trying to use the codec peer specific |
05:06.27 | Snake-Eyes | every time i setup rfc2833 nothing seems to registry on the other side eg navigating pbx menu system |
05:06.36 | SwK | iceyp: codecs listed in general set the overall codecs and if its disabled there its disabled everywhere |
05:07.01 | SwK | what ATA are you using? |
05:07.18 | SwK | what ATA with the 2833 issue that is? |
05:07.32 | Snake-Eyes | ive tried it on gxp-2000 and spa941 |
05:07.37 | iceyp | i'm using a cisco 7912, [general] disables 729 but my cisco 7940 context enables 729 |
05:07.41 | iceyp | and i can call with 729 |
05:08.19 | iceyp | maybe i broke something with peer/user/friend |
05:08.21 | Snake-Eyes | hmm, i have feeling the people doing line/pstn termination might be one reason |
05:08.45 | Snake-Eyes | i have no codecs allowed or disallowed under [general] |
05:12.06 | iceyp | weird, now it uses ulaw even though i've disallowed everything |
05:12.35 | iceyp | May 24 17:12:27 NOTICE[50174]: chan_sip.c:3646 process_sdp: No compatible codecs! |
05:12.43 | iceyp | and disallow=all in [general] |
05:12.56 | iceyp | however on the specific user i've allowed g729 & ulaw |
05:14.18 | Snake-Eyes | i 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 |
05:14.51 | iceyp | mmmm ok |
05:14.55 | iceyp | so its known |
05:15.17 | iceyp | but how come if i disallow=all in [general] it doesnt use the allowed ones in the users context? |
05:15.18 | Snake-Eyes | yea |
05:15.47 | iceyp | it didnt use to act that way |
05:16.06 | Snake-Eyes | dont know |
05:16.21 | iceyp | maybe its the type=friend vs type=peer thing now |
05:16.46 | Snake-Eyes | did you have disallow=all under users? |
05:16.58 | iceyp | yes |
05:17.08 | iceyp | and then allow=g729 allow=gsm allow=ulaw |
05:17.14 | iceyp | in that order |
05:18.05 | iceyp | when i set type=friend then it uses the codecs below that user |
05:18.26 | Snake-Eyes | im understanding is the local context (peer, friend, user) overides the general context |
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05:40.34 | Flauto | how call parking works? |
05:40.42 | Flauto | is there anyone willing to help |
05:41.51 | Strom_C | you 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) |
05:50.31 | Crshman | is there a limit as to what extensions i can use? |
05:51.03 | *** join/#asterisk Kis (i=vlad@p5080FD06.dip.t-dialin.net) |
05:51.08 | Strom_C | what do you mean, Crshman? |
05:51.18 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
05:51.24 | Crshman | like i can use 1xx 2xx 3xx 4xx with no limitations? |
05:51.27 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
05:51.35 | Strom_C | what do you mean "limitations" |
05:51.43 | Crshman | are any extensions blocked out? |
05:51.53 | drray | I don't think any extensions are blocked off by default in extensions.conf |
05:51.56 | Strom_C | what do you mean "blocked out"? |
05:52.00 | drray | features maybe |
05:52.11 | Strom_C | you can assign anything you want anywhere you want |
05:52.20 | drray | the * features might supercede |
05:52.30 | Strom_C | well sure, vertical service codes |
05:52.35 | Crshman | o |
05:52.43 | Crshman | where is a listing of those? |
05:52.48 | Strom_C | but thats why you look at www.nanpa.com to find out :) |
05:53.04 | drray | feature.conf |
05:53.06 | Strom_C | i believe you can assign your own vertical service codes in the *95-*99 range |
05:53.10 | drray | er, features.conf |
05:53.29 | *** join/#asterisk just_a_guy (n=beetle_b@74.136.209.21) |
05:53.31 | Crshman | excelent thnx |
05:53.31 | Strom_C | Crshman: leave vertical service codes to nanpa specs |
05:53.37 | Strom_C | www.nanpa.com |
05:54.09 | Strom_C | only conflict with the existing numbering plan if you know EXACTLY what you're doing |
05:54.38 | Crshman | no i'm trying not to =) |
05:54.43 | just_a_guy | Upon loading asterisk, I get the following error: [codec_speex.so]Ouch ... error while writing audio data: : Broken pipe |
05:54.52 | just_a_guy | Ideas? |
05:54.56 | drray | Crshman - I use #434 for an extension |
05:55.02 | drray | including the # sign |
05:55.27 | Crshman | ooooooooooo so not just the 4xx number? you add a #? |
05:55.39 | drray | it can be any number you want |
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05:56.02 | Crshman | ok |
05:56.39 | drray | it could be ABCD if you had a phone set that supported it |
05:58.35 | *** join/#asterisk asteriskster (n=aahmed@202.5.145.13) |
06:02.30 | Strom_C | don't use # to start an extension |
06:02.41 | Strom_C | # indicates completion of dialing |
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06:04.32 | Crshman | how much processor does software echo cancellation use? |
06:04.54 | *** join/#asterisk af_ (n=af@ip-143-220.sn1.eutelia.it) |
06:04.55 | beetle_b2 | Sorry - I lost the connection last time. Question: Running asterisk gives me the error: [codec_speex.so]Ouch ... error while writing audio data: : Broken pipe |
06:07.13 | Strom_C | anyone here have experience configuring sangoma cards on PRI circuits? |
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06:11.09 | *** part/#asterisk KaBewM (n=DA-MAN@66-215-7-106.dhcp.psdn.ca.charter.com) |
06:12.48 | *** join/#asterisk asteriskster (n=aahmed@202.5.145.13) |
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06:25.15 | Crshman | i'm having a great issue with echo on my setup.....how can i fix this? |
06:26.32 | Strom_C | Crshman: what kind of lines are you using? |
06:26.45 | Crshman | IAX out and SIP in |
06:27.03 | angler- | Crshman, not much u can do in all voip |
06:27.23 | Crshman | so there is no fix for the echo problem? |
06:27.35 | Strom_C | Crshman: what kind of telephone set are you using? |
06:28.02 | Crshman | i'm using a softphone to dial out to a cell phone and i get echo |
06:28.24 | Strom_C | Crshman: do you have a headset for the softphone, or are you using a microphone and speakers? |
06:28.47 | Crshman | mic and speakers, i have enabled the "echo cancellation" for mic and speakers in the softphone app... |
06:29.01 | Strom_C | there's a reason quality speakerphones cost hundreds of dollars |
06:29.12 | Strom_C | the echo is coming from your poor setup |
06:29.17 | Strom_C | get a headset or a real telephone |
06:29.19 | angler- | id say so too |
06:29.30 | Crshman | o ok i thought it was the actual lines |
06:29.33 | Crshman | i'll try it |
06:29.50 | Strom_C | so, any sangoma zealots out there willing to help with a PRI config issue? |
06:30.45 | angler- | maybe if it was digium pri card |
06:30.56 | drray | isn't it the same damn thing? |
06:30.56 | Strom_C | yeah, I know, I much prefer the digium cards |
06:31.04 | *** join/#asterisk lorinc (n=ang@caracas-4338.adsl.interware.hu) |
06:31.17 | Strom_C | but I've been brought in to fix this fucked up asterisk install, and the previous installer used a sangoma card |
06:31.42 | Strom_C | drray: no, the sangoma cards patch zaptel and have sixty-five extra ass-backwards config utilities |
06:32.14 | angler- | drray, I won't touch sangoma |
06:32.29 | *** join/#asterisk achandra_ (n=achandra@static-71-103-255-118.lsanca.dsl-w.verizon.net) |
06:32.37 | Strom_C | neither will I when I'm specing out the hardware - this is ridiculous |
06:32.45 | angler- | hehe |
06:33.07 | drray | that 8 port sangoma card would solve a few of my problems |
06:33.19 | Strom_C | I'm surprised - usually you start bagging on sangoma and some zealot pops out of nowhere to tell you how wonderful sangoma is |
06:33.32 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
06:33.39 | Strom_C | no such luck tonigt, I guess |
06:33.51 | achandra_ | 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 |
06:34.12 | pjchilds | (zealot mode) ooh... sangoma is wonderful... |
06:34.39 | achandra_ | the doc does have the use of flags to deal with it... |
06:34.50 | achandra_ | but im unclear on the nomenclature.. |
06:35.31 | Crshman | cool thnx folks i got it fixed using a better headset echo is nearly gone |
06:35.57 | stephane_ | jour |
06:37.49 | *** join/#asterisk supjigatr (n=syslod@152.53.16.10) |
06:38.23 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
06:39.09 | supjigatr | Anyone recommend disto that plays nice with sangoma a104d? |
06:40.00 | *** join/#asterisk nfinetin (n=nfi@LSt-Amand-152-32-9-55.w82-127.abo.wanadoo.fr) |
06:40.24 | *** join/#asterisk FuriousGeorge (n=get@ool-43536ea8.dyn.optonline.net) |
06:40.27 | FuriousGeorge | hey all |
06:40.41 | nfinetin | hi |
06:40.50 | FuriousGeorge | just wondering, i got a server im building thats gonna have 2 tdm400ps |
06:40.50 | supjigatr | Hi |
06:40.59 | FuriousGeorge | so irq sharing is a concern |
06:41.11 | Strom_C | use a tdm2400 :) |
06:41.15 | FuriousGeorge | should i get a pci or pci-e gpu? |
06:41.44 | FuriousGeorge | does it matter? |
06:43.04 | FuriousGeorge | i 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? |
06:43.35 | supjigatr | You will likely have problems. |
06:43.43 | FuriousGeorge | then again i got those little tiny pci slots. i guess those are pci 1x lemme get that manual |
06:44.59 | Strom_C | how charming! when you unload the sangoma module, the system grinds to a screeching halt |
06:45.04 | Strom_C | that's just fabulous |
06:45.48 | FuriousGeorge | ok so i got 4 pci slots, two will be used for tdms, then i got 2 pci-e X1 and 1 pci-e X16 |
06:45.50 | drray | right now I'm using a go varion.com tor2 clone, and am thinking about getting the new Sangoma 8 t1 port card |
06:45.54 | pjchilds | achandra, 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() |
06:46.50 | FuriousGeorge | does it make sense that my pci-e slots are less likely to share an irq with the other pci slots? |
06:48.47 | pjchilds | achandra, 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) |
06:50.17 | tparcina | hi group! |
06:50.30 | nfinetin | hi |
06:50.46 | Strom_C | hi! |
06:50.54 | nfinetin | is there a way to use an other prompt language ? like french or german ? |
06:51.06 | angler- | nfinetin, yup |
06:51.36 | FuriousGeorge | it 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 |
06:51.37 | tparcina | anybody uses asterisk-stat |
06:51.44 | angler- | nfinetin, Use set for the language variable to change it to a different sounds directory |
06:52.13 | tparcina | i have problem conecting to mysql database. asterisk-stat can't connect and show data from mysql |
06:52.19 | nfinetin | where can i change that variable i mean witch confile ? |
06:52.51 | tparcina | set(language(de)) |
06:53.10 | tparcina | nfinetin, something like that - check on voip-info |
06:53.38 | tparcina | look for application - set |
06:54.17 | tparcina | so, anybody knows how to check why asterisk-stat doesn't connect to mysql database? |
06:54.31 | nfinetin | ok thx, an other thing is that i cannot figure out how to run rxfax and txfax according to docs |
06:55.18 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
06:55.31 | tparcina | never used rxfax and txfax, sorry |
06:56.04 | angler- | nfinetin, those apps are apart of spandsp, the docs on their site seem pretty straight forward |
06:56.35 | nfinetin | is there something else that i could use for faxes on * |
06:56.49 | angler- | depends |
06:56.52 | nfinetin | runnning with freepbx |
06:57.25 | angler- | well i would stick with spandsp |
06:57.43 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
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06:58.19 | Zeeek | ManxPower ? |
06:58.21 | CrashHD | if a call is sent to a queue with QUEUE() and no members are avilable and joinempty = strict? |
06:58.29 | CrashHD | what happens to the call? |
06:59.44 | Strom_C | CrashHD: dead hookers |
06:59.58 | angler- | lol |
07:00.00 | angler- | what? |
07:00.27 | CrashHD | lol |
07:00.29 | CrashHD | off the wall |
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07:07.28 | Crshman | what is a good TTS program? |
07:07.31 | *** join/#asterisk satlan32 (n=pargit@212.150.142.211) |
07:07.45 | satlan32 | good morning.. |
07:07.59 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43.pool8172.interbusiness.it) |
07:08.05 | satlan32 | good morning.. |
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07:19.35 | *** join/#asterisk fugitivo (n=ajf@190.48.166.195) |
07:19.37 | fugitivo | hi |
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07:21.07 | *** part/#asterisk achandra_ (n=achandra@static-71-103-255-118.lsanca.dsl-w.verizon.net) |
07:21.11 | satlan32 | hi |
07:21.22 | satlan32 | anyone used audiocodes mp 104 fxo? |
07:21.47 | Zeeek | hi |
07:21.49 | satlan32 | i 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) |
07:23.29 | *** join/#asterisk Assid (n=assid@203.115.83.214) |
07:23.36 | Assid | heya |
07:25.19 | Assid | whats the best way to connect a legacy pbx into voip |
07:26.01 | *** join/#asterisk RestLessGemini (n=rLg@202.141.247.150) |
07:26.13 | satlan32 | voip gateway |
07:26.30 | satlan32 | such as digium cards, audiocodes media gw |
07:27.09 | Assid | well.. i was thinking of digium cards.. but.. i keep getting mixed up between fxo and fxs |
07:27.16 | Assid | fxo is where you dont procduce dialtone right |
07:28.44 | Assid | hey the linksys pap2 would do as well right ? |
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07:53.06 | nfinetin | hye again, i've figure out how to install sapndsp it was a patching issue |
07:53.11 | nfinetin | thx for your help |
07:53.54 | nfinetin | why do i get line twice times in the feature vcore admin modules |
07:54.29 | angler- | thats probably a question for #freepbx |
07:55.23 | nfinetin | ok sorry wrong window |
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08:02.48 | bartlebee | could anyone have cisco ip phone 7960 firmware ver 6.x they're willing to share please? |
08:02.54 | bartlebee | could = would |
08:03.55 | x86 | try #cisco |
08:06.07 | bartlebee | k, waiting on #cisco to be directed to a url and contact your nearest helpful cisco dealer |
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08:11.40 | MrChimpy | hey *ers |
08:13.08 | bartlebee | if we tell you how will you let MrChimpy know? |
08:19.05 | stephane_ | re |
08:19.48 | *** join/#asterisk Tusker (n=tusker@203.117.94.152) |
08:19.55 | bartlebee | MrChimpy :) |
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08:22.19 | Tusker | heya 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? :) |
08:22.45 | Tusker | oh... and asterisk is responding 'SIP/2.0 404 Not Found' |
08:29.03 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
08:31.59 | MrChimpy | why have one interface when you can have three different ones? |
08:32.12 | angler- | Tusker, should be looking for an extension in the context |
08:32.37 | Tusker | angler-: ahh i see... rather than s, just specify blah-user... thanks! |
08:33.43 | Tusker | ok, another question... 'Their Codec Capability: 271' < how do I know what formats the 271 mask is ? |
08:34.31 | angler- | look at sip debug, it will tell what each codec is |
08:35.32 | x86 | Tusker: 'show codecs' from the CLI |
08:35.58 | Tusker | ahhh very nice |
08:36.01 | Tusker | show codec 271 |
08:36.34 | angler- | 271 is a combination of several codecs |
08:36.50 | Tusker | yeah, correct, show codec parses the mask |
08:41.27 | Tusker | ok, 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 ? |
08:43.31 | angler- | r option on dial |
08:45.48 | Tusker | angler-: even after it has done the call establish? |
08:46.32 | Tusker | ie, ring while "silence detected" |
08:46.33 | *** join/#asterisk jerlique (n=jerlique@lnk6.adl5.adsl.esc.net.au) |
08:47.12 | *** join/#asterisk somegeek (i=levin@unaffiliated/somegeek) |
08:47.20 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
08:52.13 | *** join/#asterisk littlejohn (n=little@host221-76.pool876.interbusiness.it) |
08:52.49 | Tusker | btw, is it worth to purchase the g729 codec ? |
08:55.24 | Zeeek | how many channels? |
08:56.04 | Tusker | 1 to 2 channels I think should be enough |
08:56.16 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
08:57.00 | Zeeek | for $20 take the plunge ! |
08:57.36 | Tusker | is it legal to "try" it before buying? |
08:57.47 | Zeeek | a 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) |
08:57.58 | Tusker | :) |
08:58.20 | Zeeek | not as far as I know |
08:58.30 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
08:58.39 | Tusker | ok |
08:59.10 | Zeeek | I've watched the bandwidth, it definitely saves it and it definitely sounds good |
08:59.21 | Zeeek | but not as good as ulaw IMO |
08:59.50 | Tusker | ah ok |
09:00.11 | coppice | the main reason to use G.729 is "the other end requires it" |
09:00.27 | Zeeek | good point |
09:00.51 | Tusker | but, if the other end has ulaw, and bandwidth isn't a huge issue, then that should be fine ? |
09:01.22 | coppice | if bandwidth is not an issue, then ulaw/Alaw is a better choice. |
09:01.30 | Zeeek | as I said :) |
09:02.08 | Tusker | ok, cool then |
09:04.47 | *** part/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl) |
09:04.52 | *** join/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl) |
09:05.02 | *** join/#asterisk mr_horsepower (n=igor@82.102.1.42) |
09:17.44 | Assid | isnt ulaw lossy ? |
09:17.54 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
09:18.48 | x86 | ulaw is lossless |
09:18.54 | x86 | same with alaw |
09:19.06 | x86 | that's how it is possible to do fax with them ;) |
09:19.39 | coppice | ulaw and alaw are lossy, but only a little bit |
09:20.20 | x86 | no? |
09:20.56 | Assid | well .. 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) |
09:22.37 | x86 | Assid: could be from latency |
09:22.54 | x86 | Assid: also could be that they can not handle a 64kbps stream of audio |
09:23.04 | x86 | Assid: try switching to GSM |
09:23.11 | x86 | (if the clients support it) |
09:23.16 | x86 | else, try g726 |
09:23.42 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
09:23.44 | Assid | well.. what i did is gsm from/to provider (voicepulse) and ulaw for internal communication |
09:23.56 | Assid | but i wonder what happens in a scenario where the calls end up in meetme |
09:25.50 | x86 | Assid: "internal" == LAN? |
09:27.05 | coppice | Assid: 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) |
09:27.50 | Assid | well.. people from the outside.. get broken audio.. |
09:27.54 | Assid | or their audio breaks |
09:27.59 | Assid | lan people got no issues |
09:28.09 | Assid | HOWEVER.. there is ample bandwith still left |
09:28.38 | Assid | on another note.. is it possible to link to meetme locations? |
09:28.41 | coppice | you probably need QoS. no amount of bandwidth is adequate all the time without a little management |
09:28.48 | Assid | like meetme on box1 and meetme on box2 ? |
09:33.16 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
09:33.31 | *** part/#asterisk sshadow (n=sshadow@213-84-101-107.adsl.xs4all.nl) |
09:33.33 | Assid | brb.. need food |
09:34.06 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
09:34.06 | *** mode/#asterisk [+o denon] by ChanServ |
09:36.05 | *** part/#asterisk RestLessGemini (n=rLg@202.141.247.150) |
09:36.33 | x86 | Assid: 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) |
09:36.52 | x86 | Assid: what your problem most likely is, is latency between you and your VoIP provider |
09:37.03 | x86 | Assid: also, could be having jitter issues |
09:37.05 | coppice | x86: are you really this thick, or just trying to be annoying? |
09:37.23 | x86 | thank god for ignore :) |
09:38.27 | x86 | Assid: 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) |
09:39.15 | Tusker | x86: if there are jitter issues, what can be done about it ? |
09:39.28 | x86 | Tusker: you can increase your buffers usually |
09:39.45 | Tusker | how would I go about doing that? :) |
09:39.52 | x86 | Tusker: 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) |
09:40.20 | x86 | Tusker: depends on how you are connecting to your provider (TDM, IAX, SIP, whatever) |
09:40.32 | Tusker | x86: SIP |
09:40.35 | x86 | Tusker: if you're using a zaptel card, you can change some stuff in zapata.conf |
09:41.27 | Tusker | no zaptel card |
09:41.45 | x86 | http://www.voip-info.org/tiki-index.php?page=Asterisk+new+jitterbuffer |
09:41.55 | coppice | Tusker: the good jitter buffering in * is currently only for IAX. onyl recent test versions have good jitter handling for SIP |
09:42.27 | Tusker | oh ok, so best to use IAX where possible ? |
09:42.32 | coppice | s/in */in releases of */ |
09:43.21 | coppice | or wait for 1.4 :-) |
09:43.42 | Tusker | how much regex does jbot know ? |
09:44.21 | Tusker | s/\s(\w+)/_\s_\1/g |
09:45.02 | coppice | i think it just does simoke substitutions when it sees the s// |
09:45.05 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.179) |
09:45.28 | Tusker | ok, cool though |
09:47.28 | Zeeek | jbot knows how to run asterisk through NAT :) |
09:47.51 | coppice | and make coffee and doughnuts |
09:47.53 | Zeeek | jbot receives faxes in any codec |
09:48.13 | Tusker | wow, sounds decent |
09:48.20 | Zeeek | jbot servers beer to all, but knows to serve lite to those with a weight problem |
09:48.44 | Zeeek | s/servers/serves/ |
09:48.45 | *** join/#asterisk buzzyd (n=buzzyd@82-45-247-173.cable.ubr01.enfi.blueyonder.co.uk) |
09:49.06 | coppice | does it distinguish those who are fat, but don't have a problem with that? |
09:49.14 | Zeeek | absolutely! |
09:50.01 | buzzyd | I 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? |
09:50.11 | mitcheloc | is jbot an addon for asterisk? can someone link me...? |
09:51.08 | buzzyd | Can I supply information that will help troubleshoot or is their a better place for me to ask this question? |
09:51.42 | Zeeek | put your nonces in the fridge when not in use, they're STALE! |
09:52.32 | RoyK | Zeeek: fridge is full... |
09:52.34 | *** join/#asterisk nfinetin (n=nfi@LSt-Amand-152-32-9-55.w82-127.abo.wanadoo.fr) |
09:52.43 | buzzyd | Zeeek, my fridge is very full |
09:52.50 | Zeeek | remove the useless stale credentials |
09:53.02 | RoyK | litterbox is full as well |
09:53.07 | Zeeek | for bad password errors, make them GOOD |
09:53.11 | coppice | RoyK: 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) |
09:53.24 | buzzyd | Zeeek, how do I remove stale credentials |
09:53.34 | Zeeek | it's actually about 12°C here now, very cold for the season |
09:54.07 | RoyK | 15 degrees and sun |
09:54.18 | buzzyd | is this an asterisk problem or snom? |
09:54.18 | coppice | if it makes you feel any better we aren't all suffering that :-) |
09:54.41 | Zeeek | buzzyd try a softphone to check that |
09:55.10 | buzzyd | soft phone works fine as does my snom 190 and the other 320 I've been testing with |
09:55.29 | buzzyd | what is confusing me is I don't know what a nonce is |
09:55.41 | RoyK | buzzyd: a hash |
09:55.43 | buzzyd | how/where its generated |
09:55.43 | Zeeek | try googling for "stale nonce" |
09:55.47 | RoyK | buzzyd: a joint, perhaps |
09:55.54 | buzzyd | would love one :) |
09:55.58 | Zeeek | as in "see you anonce..." |
09:56.09 | Makenshi | a nonce is an english word for insulting someone |
09:56.17 | Makenshi | "you nonce!" |
09:56.26 | Zeeek | but seriously GIYF - try looking there |
09:56.39 | Zeeek | the second link is "what is a stale nonce?" |
09:56.52 | RoyK | for the nonce => for the present; temporarily |
09:56.55 | Zeeek | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg114348.html |
09:57.20 | buzzyd | looking now but according to description that shouldn't affect registration |
09:57.20 | RoyK | "Ubuntu: Ancient African word for ''I'm sick of compiling Gentoo''" -- Jeff Waugh |
09:57.45 | Zeeek | Olle says the message is just a warning |
09:57.59 | Zeeek | glad I took the time to look |
09:58.23 | coppice | sagoma - 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) |
09:58.55 | Zeeek | sang=blood |
09:59.02 | buzzyd | anyone here use snom 320's? |
09:59.14 | Zeeek | nah, too many stale nonces in them |
09:59.19 | buzzyd | lol |
09:59.21 | RoyK | Zeeek: sang == song :P |
09:59.28 | coppice | its sangoma, not sanguine |
09:59.34 | Zeeek | they're supposed to be great for the money though :) |
09:59.41 | coppice | you much be thinking of MS |
10:00.00 | Zeeek | Multiple Suckezrs? |
10:00.16 | mut | i like my sangoma a104d, sorta |
10:00.18 | buzzyd | Zeeek, the fact it ain't login in would suggest the opposite :) |
10:00.20 | mut | it broke my faxing tho |
10:00.42 | coppice | the sangoma cards are nice, but the installation needs a serious overhaul |
10:01.01 | Zeeek | buzzyd could be 2 different problems |
10:01.02 | *** join/#asterisk Vyeperman (n=Vye@ip68-6-130-59.sd.sd.cox.net) |
10:01.52 | buzzyd | Zeeek, please elaborate |
10:01.52 | Zeeek | looked at the wiki? Some phones have install intructions for asterisk there |
10:01.58 | *** join/#asterisk littlejohn (n=little@host221-76.pool876.interbusiness.it) |
10:02.15 | Zeeek | I mean maybe the nonce is a warning having naught to do with not logging in |
10:04.09 | znoG | has anyone used the clipcomm ATAs? |
10:04.57 | coppice | http://www.voip-info.org/wiki/view/Clipcomm+ATA+and+Gateways |
10:05.56 | buzzyd | Zeeek, 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? |
10:07.17 | Zeeek | yes, if you compared it it sounds that way. Next step, find a config somewhere |
10:07.33 | Zeeek | and look at sip debug |
10:07.52 | *** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net) |
10:09.05 | buzzyd | Zeeek, Ok thanks for pointers off to do some more testing |
10:09.42 | znoG | coppice: 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) |
10:11.09 | Tusker | is 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 ? |
10:12.13 | znoG | coppice: 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) |
10:17.24 | acrg | I'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? |
10:19.38 | coppice | acrg: 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) |
10:21.01 | acrg | thanks |
10:21.16 | edo1 | hi |
10:21.52 | edo1 | exist ISDN COLP (Connected Line Identification Presentation) support in asterisk? |
10:22.40 | *** join/#asterisk Ecio (n=eciostar@194.105.59.42) |
10:23.01 | acrg | edo CLI? yes |
10:24.28 | *** join/#asterisk pbx1 (n=pbx1@58.69.229.213) |
10:25.13 | edo1 | no, not cli |
10:26.32 | edo1 | example - 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) |
10:31.54 | edo1 | for outgoing calls i can set callerid, what for incoming? |
10:32.49 | *** join/#asterisk xorol (n=root@uu212-190-229-201.unknown.uunet.be) |
10:33.02 | qdk | edo1: you wanna |
10:33.04 | qdk | argh |
10:33.18 | qdk | edo1: you wanna SET callerid on incoming calls? |
10:37.02 | tzafrir | edo1, 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) |
10:38.40 | eset | anyone 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) |
10:40.05 | Ecio | hi all, im experienced bad (well, let's say not-so-good) quality with asterisk, can someone give me some hint? |
10:40.12 | puzzled | hi |
10:40.21 | *** join/#asterisk xorol (n=dannyz@uu212-190-229-201.unknown.uunet.be) |
10:40.55 | dtwilson | Ecio: you mean poor audio quality on phones? |
10:41.07 | dtwilson | Ecio: are you using sp handsets? |
10:41.08 | Ecio | im 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 |
10:41.10 | dtwilson | sip* |
10:41.25 | Splat | anyone know how well the cisco 7940 ip phones work with asterisk? |
10:41.37 | Ecio | i.e. if i try to call with x-lite (directly connected to asterisk with a sip account) |
10:41.43 | Ecio | and call echo test (or the conference) |
10:41.54 | *** join/#asterisk zotz (n=zotz@24.231.36.9) |
10:42.19 | Ecio | the 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 |
10:42.35 | drray | Splat - I use a 7960 cisco |
10:42.40 | znoG | the 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) |
10:42.45 | dtwilson | Ecio: this may sound strange - but do you have power cables running parallel to the relevant network cables in close proximity? i.e. tied together? |
10:44.01 | Ecio | dtwilson: 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 |
10:44.28 | Ecio | i've also downloaded the audio files from astlinux.com (that afaik are supposed to be better quality) but it doesnt change |
10:44.39 | Splat | drray: that doesn't help.. the place I have to do some work has 7940's on every desk.. |
10:44.49 | Ecio | initially i thought it was maybe a problem of timers on linux and vmware... but i have problems also on the physical machine too |
10:45.03 | Ecio | consider 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) |
10:45.49 | drray | Splat - 7940's and 60's are the same cept the number of lines |
10:45.54 | Ecio | i've found also italian voice on tomato.it but i have this "rumors" on those too |
10:46.25 | Ecio | calling 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...) |
10:47.26 | coppice | Ecio: 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 |
10:47.26 | Assid | is 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) |
10:48.53 | dtwilson | coppice: 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? |
10:49.04 | Ecio | coppice : uhm... let me check it |
10:49.19 | coppice | Assid: have you tried? it should kinda work, but I suspect the way mixing occurs might give some quality issues |
10:49.21 | Ecio | do u remember the hdparm syntax for checkin it' |
10:50.03 | Ecio | using hdparm -d /dev/hda i got using_dma = 1 (on) |
10:50.22 | coppice | dtwilcon: 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) |
10:50.27 | MrChimpy | my 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 |
10:51.01 | Ecio | UDMA modes: udma0 udma1 udma2 udma3 udma4 *udma5 <- * = active mode |
10:51.09 | *** join/#asterisk Hadaka (i=naked@naked.iki.fi) |
10:51.13 | Ecio | so prolly it's not dma :/ |
10:51.25 | MrChimpy | i 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 |
10:51.35 | Splat | drray: ok, so they work fine then? any tricks to them or anything? |
10:51.56 | Hadaka | Hello, 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? |
10:53.40 | Hadaka | I know the reverse is possible (connecting with SIP to asterisk and having it use IAX outbound), but I don't know about this need? |
10:55.10 | drray | Splat - they work great, once setup for sip, you need a TFTP server for them and a smartnet contract |
10:55.35 | Splat | drray: smartnet contract? |
10:56.31 | Ecio | have to go to dinner... later |
10:57.01 | drray | Splat - for getting firmware for the phones |
10:57.37 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
10:57.43 | Splat | drray: 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) |
10:57.55 | supjigatr | Hi. |
10:58.13 | drray | Splat _ I'd assume so |
10:58.17 | dpryo | Hadaka: That shouldn't be a problem. |
10:58.21 | drray | Splat - or maybe just one contract :) |
10:58.26 | supjigatr | Anyone know how to recover IAXy boxes after reinstall of a asterisk server without bringing them back in. |
10:58.26 | Hadaka | dpryo: Great! |
11:02.37 | Splat | drray: 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 |
11:02.58 | drray | Splat - it depends on what they want |
11:03.46 | edo1 | tzafrir: i try without bristuff, plain asterisk |
11:04.05 | Splat | need 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 |
11:05.14 | MrChimpy | dammit, there's an AMI GetVar as it is. |
11:05.21 | *** join/#asterisk rkr245 (n=ravi@81.21.33.35) |
11:09.38 | *** join/#asterisk dtwilson (n=dave@host217-36-121-129.in-addr.btopenworld.com) |
11:16.31 | satlan32 | hi |
11:16.36 | satlan32 | question not regarding asterisk |
11:16.50 | satlan32 | tcpdump -i eth0 -s 1500 |
11:16.59 | satlan32 | how do i set it up to save to a file? |
11:17.23 | *** join/#asterisk RoyKa (n=roy@213.160.242.91) |
11:21.30 | dtwilson | satlan32: just append '> filename.txt' |
11:21.45 | satlan32 | thanks |
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11:24.36 | *** join/#asterisk muppetmaster (n=MuppetMa@81.184.73.169) |
11:25.50 | muppetmaster | Hello |
11:30.48 | *** join/#asterisk I-MOD (i=opticron@68.62.165.168) |
11:32.27 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
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11:37.03 | motu | which 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 |
11:38.55 | *** join/#asterisk muppetmaster (n=jasongoe@169.red-81-184-73.user.auna.net) |
11:39.04 | Henk | I 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) |
12:00.20 | *** join/#asterisk stoffell_h (n=PircBot@pot.catsanddogs.com) |
12:01.03 | *** join/#asterisk tparcina (n=tparcina@wr-lama.iskon.hr) |
12:01.49 | tparcina | call file, - why this one doesn't call? - http://pastebin.ca/58619 |
12:02.15 | motu | I get Ignoring hanguponpolarityswitch, and Ignoring signalling, why? |
12:02.24 | tparcina | i 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) |
12:08.05 | *** join/#asterisk aBd0ulaX (n=abd@LNeuilly-152-21-121-86.w193-253.abo.wanadoo.fr) |
12:09.39 | [TK]D-Fender | tparcina : 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) |
12:11.38 | fugitivo | hi |
12:11.38 | aBd0ulaX | Hello, 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 |
12:11.48 | fugitivo | ~docs |
12:11.55 | jbot | i 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 |
12:12.24 | aBd0ulaX | thx man ;) |
12:13.13 | fugitivo | np |
12:13.27 | Zeeek | aBd0ulaX there are several French language sites about asterisk |
12:13.44 | Zeeek | french-asterisk.net or something like it. Google to find them |
12:13.53 | Ahrimanes | Zeeek: :) |
12:13.59 | Zeeek | Beeer |
12:14.23 | aBd0ulaX | cool ;) |
12:14.30 | Zeeek | n'est-ce pas? |
12:14.47 | aBd0ulaX | haha t français ?? |
12:14.51 | Zeeek | directcentrex and wengo use asterisk |
12:15.07 | Zeeek | also Acropolis Télécom |
12:15.15 | fugitivo | i'm having serious problems with a te205p, my zttest is really bad |
12:15.17 | Zeeek | Axialys |
12:15.18 | fugitivo | i tried everything |
12:15.33 | Zeeek | fugitivo cheat and you'll pass :) |
12:15.50 | fugitivo | and the motherboard is not in the incompatibility list |
12:16.02 | *** join/#asterisk lorinc (n=ang@caracas-4553.adsl.interware.hu) |
12:16.05 | Ahrimanes | fugitivo: minor issues |
12:16.32 | fugitivo | Best: 100.000000 -- Worst: -79.895020 -- Average: 98.767997 |
12:16.39 | Ecio | more 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? |
12:16.40 | fugitivo | that's not a minor issue |
12:17.01 | Zeeek | Ecio are you using a cell phone? |
12:17.07 | Ecio | no: x-lite |
12:17.11 | fugitivo | and digium support doesn't reply emails |
12:17.16 | Zeeek | make sure xmit silence is off |
12:17.30 | Zeeek | fugitivo call digium |
12:17.34 | Zeeek | in a few hours |
12:17.53 | fugitivo | i sent 3 emails yesterday |
12:18.02 | Zeeek | what country are you in? |
12:18.11 | iDunno | is it nearly home time yet? |
12:18.14 | fugitivo | Argentina |
12:18.29 | Ecio | zeek xmit silence is off |
12:18.30 | Zeeek | I was afraid of that fugitivo, ar is blocked on many mail servers |
12:18.37 | fugitivo | ??? |
12:18.47 | Zeeek | try writing from a yahoo web mail or something |
12:18.59 | fugitivo | they did send the automatic reply |
12:19.02 | Zeeek | fugitivo you didn't know AR is one of the capitals of spam? |
12:19.15 | Zeeek | ok if you got that the mail came thru |
12:19.58 | fugitivo | Zeeek: i don't think so, in AR we don't have enough bandwidth ;)\ |
12:20.45 | Zeeek | I guarantee you a lot of spam comes from there. Not as much as China, but a lot |
12:21.21 | Zeeek | Brazil too, while we're putting down south america ;) |
12:21.25 | Ecio | zeek: im tryin also from my cisco phone (via sip trunk phone ->callmanager -> asterisk conference) |
12:21.35 | Ecio | and music disappears after some seconds.... |
12:22.06 | Zeeek | Ecio all I know is that behavior is common on cell phones, I gues they use silence suppression for obvious reasons |
12:22.28 | Zeeek | X-Lite should work so something is indeed wrong |
12:24.58 | sudhir492 | SOS - Anyone using spandsp and app_rxfax here? |
12:25.06 | Ecio | zeek: 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.. |
12:26.00 | Zeeek | sudhir492 yes. rx works great sometimes, other times it won't receive a fax here |
12:26.20 | mitcheloc | isn't asterfax the replacement? |
12:26.38 | sudhir492 | Zeeek: Can you tell which version of spandsp and rxfax are you using? |
12:26.43 | dtwilson | Ecio: 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) |
12:27.18 | Zeeek | sudhir492 asterisk 1.2 and whatever the latest spandsp was at the time |
12:27.26 | Zeeek | let me login and see |
12:27.30 | Ecio | dtwilson: 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...) |
12:27.32 | sudhir492 | ok |
12:27.46 | Ecio | im 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) |
12:28.12 | sudhir492 | I 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) |
12:28.34 | sudhir492 | but get immediate hangup on the Zap channel |
12:28.46 | Zeeek | looks like 2pre21 I have |
12:29.01 | sudhir492 | what about rxfax |
12:29.19 | Ecio | dtwilson: 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 |
12:30.29 | Ecio | that's really strange... but the worst problem is the one i wrote some hours ago, that recorded messages dont play very well... |
12:30.29 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
12:33.10 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43.pool8172.interbusiness.it) |
12:33.10 | Zeeek | I am not easily finding rxfax version |
12:33.10 | Zeeek | (or source) |
12:33.11 | dtwilson | Ecio: You recorded messages *from* a handset or xlite etc? My guess for those the problem might simply be poor quality microphones |
12:33.31 | Zeeek | Ecio have you checked for IRQ confilcts? |
12:33.31 | Ecio | dtwilson: sorry i didnt explain myself, i meant the announcement |
12:33.32 | Ecio | like "this is echo test " or "this is a conference etc.." |
12:33.33 | Zeeek | if you mean sound is jumping around and uneven it's often IRQ |
12:33.35 | Ecio | "insert the conf. number and pound key etc.." |
12:33.35 | dtwilson | ahhh the room entry announcements? -gotcha now |
12:33.38 | dtwilson | I was confused :) |
12:33.47 | Ecio | dtw: sorry, bad english :) |
12:34.42 | dtwilson | no probs Ecio :) |
12:34.49 | sudhir492 | Zeeek: I see spandsp-0.0.2pre21.tar.gz |
12:35.11 | Zeeek | that's what I have. app_rxfax.c is dated Nov 2005 |
12:35.25 | sudhir492 | I am going to try the same version. Will you please email me app_rxfax.c and app_txfax.c |
12:35.37 | Zeeek | are they no longer on the site? |
12:35.40 | *** join/#asterisk RoyKa (n=roy@213.160.242.91) |
12:35.41 | sudhir492 | my email address is sudhir492@gmail.com |
12:35.47 | dtwilson | unfortunately I'm stumped now though Ecio - I was earlier presuming you mewant you had echo problems with voice audio |
12:35.57 | dtwilson | meant* |
12:36.11 | Ecio | zeeek: the jumping audio is heard also in the cisco phone (connected to call manager) |
12:36.20 | Zeeek | IRQ? |
12:36.32 | Zeeek | sudhir492 I don't have or use txfax |
12:36.51 | sudhir492 | thats fine. rxfax is the important one for me |
12:38.22 | Zeeek | isn't it on the site? It isn't easy for me to mail it right now |
12:39.30 | Zeeek | <PROTECTED> |
12:40.21 | Ecio | zeeek 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... |
12:40.29 | eset | anyone had experience installing asterisk from source on a debian 2.6 kernel? |
12:40.38 | Zeeek | HAVE YOU CHECKED IRQ ? |
12:40.46 | eset | having trouble with the ztdummy driver |
12:41.41 | dtwilson | Ecio: check for IRQ problems on the server as per Zeeek's suggestion |
12:41.54 | Ecio | zeeek: 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-Fender | Zeeek : They're there... |
12:43.15 | Zeeek | IRQ is physical |
12:43.33 | Zeeek | so cat /proc/interrupts and see |
12:43.50 | Zeeek | yes, mine are too [TK]D-Fender |
12:44.03 | Zeeek | ANd they be lookin' real good! |
12:44.11 | Ecio | zeeek: btw i dont have any digium card... |
12:44.39 | Zeeek | ok, that never occurred to me :) |
12:45.19 | Zeeek | assuming there are no IRQ problems with ethernet (and assuming there is an ethernet interface) that would not be IRQ |
12:45.34 | Ecio | http://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 ? |
12:46.21 | Ecio | zeeek: 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. |
12:46.55 | dtwilson | airwolf: you tried bindaddress=ip.ad.dr.ess ? |
12:46.55 | [Airwolf] | What disables any communication ofcourse |
12:47.13 | Zeeek | dtwilson way too simple and logical |
12:47.17 | dtwilson | in 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) |
12:49.01 | Ecio | zeeek: 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... |
12:49.26 | Zeeek | Ecio I'm sorry, no time for that |
12:49.38 | Ecio | ok np |
12:51.10 | *** part/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
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12:57.18 | dtwilson | airwolf: 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 |
12:57.28 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
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13:00.28 | *** join/#asterisk protocoldoug (n=doug@69-160-165-210.sbtnvt.adelphia.net) |
13:00.53 | protocoldoug | for three way calling, do you use the MeetMe() command in your dial plan? |
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13:06.47 | asteriskster | cisco 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 |
13:07.15 | drray | asteriskster - why did you upgrade to 8.2? |
13:07.33 | drray | I'm just curious |
13:07.57 | asteriskster | actually 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) |
13:09.27 | asteriskster | do u know any firmware which is stable and reliable on heavy load 150-200 calls per day |
13:09.41 | *** join/#asterisk zotz (n=zotz@24.231.36.9) |
13:09.59 | *** join/#asterisk DarthClue (i=DarthClu@li14-201.members.linode.com) |
13:11.32 | dpryo | Are there any easy ways to inject audio on open channels? (Programaticly..) |
13:14.30 | asteriskster | can anyone have stable cisco 7960 SIP firmware works in heavy load to share with me |
13:15.18 | docelmo | asteriskster 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. |
13:15.38 | asteriskster | ok fine |
13:15.42 | asteriskster | sorry |
13:15.48 | asteriskster | if u mind |
13:17.52 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
13:19.10 | *** join/#asterisk noky (n=noky@200.69.211.18) |
13:19.11 | noky | hi |
13:20.29 | JackEStorm | baah, this sucks, stupid ass queue |
13:20.30 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.235.155.Dial1.SanJose1.Level3.net) |
13:21.13 | *** join/#asterisk jake1932 (n=Administ@pool-68-236-24-98.phil.east.verizon.net) |
13:21.32 | Assid | there has to be a way to link 2 meetme conference rooms from 2 different boxes together |
13:21.42 | [TK]D-Fender | JackEStorm : 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) |
13:21.54 | noky | [TK]D-Fender: hi |
13:21.57 | noky | http://pastebin.com/734977 |
13:22.12 | [TK]D-Fender | Assid : Issue a call-file for it |
13:22.12 | noky | i have realtime for extensions and sip_buddies implemented in my asterisk |
13:22.38 | noky | but i don't know why appears some querys to sql repeated..... |
13:22.51 | JackEStorm | [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. |
13:22.56 | Assid | hrmm.. that could be a good way to do it actually |
13:23.33 | [TK]D-Fender | JackEStorm : set them to no-call-waiting, and 1 call limit on the phone level. |
13:23.54 | noky | is common ? |
13:24.18 | [TK]D-Fender | noky : Bridging MeetMe's? Doubtful |
13:24.56 | noky | i have meetme... but i aren't using now... |
13:24.58 | *** join/#asterisk mercestes (n=merceste@69.15.174.114) |
13:25.02 | Assid | noky: its a very good way to cut down bandwith usage tho |
13:25.40 | JackEStorm | [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-Fender | The 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. |
13:26.26 | Assid | imagine 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-Fender | JackEStorm : 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-Fender | Assid : 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. |
13:28.20 | fugitivo | [TK]D-Fender: i couldn't solve the problem :( |
13:28.40 | fugitivo | [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) |
13:28.51 | Assid | yes, but if that person needs his phone.. for whateevr reason. we are using up the lines |
13:29.20 | Assid | also phone level versus pbx level, im guessing you might get more quality output with the pbx's calling each other |
13:30.07 | Assid | just 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-Fender | fugitivo : You know where to go now.... |
13:30.59 | Assid | or.. 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-Fender | Assid : 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) |
13:31.46 | Assid | phone gets released? |
13:32.08 | [TK]D-Fender | Assid : the 2 calls get bridged by * after that. Its in the more recent polycom firmware. |
13:32.24 | Assid | wouldnt hhe need to use the conference facility of the phone for it to work? |
13:32.27 | [TK]D-Fender | Read your Release Notes! |
13:32.49 | *** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
13:33.27 | [TK]D-Fender | Assid : 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 |
13:34.03 | Assid | damn.. i thought if you hand up your phone, the conference dies |
13:34.05 | *** join/#asterisk Inkubot (n=inkubot@200.119.229.247) |
13:34.08 | Assid | hang even |
13:34.34 | Inkubot | hi |
13:34.41 | Assid | even if its re-invite |
13:34.43 | Inkubot | i'm having a problem with my sip trunk |
13:34.44 | [TK]D-Fender | Assid : You aren't listening -> WITH THE MORE RECENT FIRMWARE <- |
13:34.56 | docelmo | hiya! |
13:34.56 | Inkubot | when i'm recieving sip calls trough a SIP trunk, i get this |
13:35.10 | [TK]D-Fender | Assid : READ THE RELEASE NOTES! They are just for starting fires you know! |
13:35.12 | Inkubot | May 24 10:35:59 NOTICE[30544] chan_sip.c: Failed to authenticate user "5629582667" <sip:5629582667@200.74.178.XXX>;tag=as1ad15b75 |
13:35.13 | Inkubot | 5629582667 is the number that calls |
13:35.20 | Inkubot | why my Asterisk try to authenticate this number as a user ? |
13:35.30 | Assid | so how would you kill the "conference" the phone starts? |
13:36.15 | [TK]D-Fender | Assid : you don't. The remain permanently bridged. A meetMe admin may be be able to kick the link if its the only one left. |
13:36.15 | Assid | also wouldnt this only work for SIP based OR if the same phone registers with BOTH the boxes |
13:36.43 | [TK]D-Fender | Assid : wouldn't matter which... |
13:37.25 | Assid | currently i have sip phones -> * <-> IAX <-> * <-> SIP phones |
13:37.44 | Assid | phone on A .. would need to dial through local * box to jump to the other box |
13:38.10 | Ecio | dtwilson & 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) |
13:38.14 | Assid | i think i need to learn more on reinvite |
13:38.26 | vgster | should sip debug peer XXX debug all sip traffic? |
13:38.31 | vgster | cos it is on my system |
13:39.09 | eset | is there anyway to run meeting rooms without a zaptel module? |
13:39.25 | vgster | ah ignore me figured it |
13:39.32 | MrChimpy | ztdummy? |
13:39.55 | eset | thats a zaptel module isnt it? |
13:39.56 | [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. |
13:40.38 | eset | i am having probs getting ztdummy to work on debian |
13:41.13 | Assid | i see |
13:41.46 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
13:42.46 | Inkubot | ejejje i solve the problem |
13:42.51 | Inkubot | insecure=very |
13:43.03 | Inkubot | : ) |
13:45.03 | Assid | just 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? |
13:45.29 | Inkubot | if a good thing to have insecure=very ? |
13:45.33 | Inkubot | if/is |
13:46.14 | *** join/#asterisk C4T3l (n=rcall01@216.54.143.2) |
13:46.21 | Ariel_ | morning everyone |
13:46.57 | [TK]D-Fender | Assid : "preference"? |
13:47.24 | docelmo | ARIEL! |
13:47.28 | docelmo | :) |
13:47.43 | docelmo | God I am so bored.. |
13:48.26 | noky | ?? |
13:48.32 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
13:48.32 | *** mode/#asterisk [+o anthm] by ChanServ |
13:48.32 | noky | http://pastebin.com/734977 |
13:48.33 | noky | :( |
13:48.33 | Ariel_ | docelm0, sorry to hear it. |
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13:49.43 | Assid | [TK]D-Fender: doesnt canreinvite get affected if both parties behind nat? |
13:49.46 | eset | wonder 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-Fender | docelm0 : 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-Fender | Assid : 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. |
13:50.11 | Ariel_ | noky, sorry your using real time. I don't know or use it. |
13:50.15 | [TK]D-Fender | Assid : Read the last sentence. |
13:50.22 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
13:50.22 | *** mode/#asterisk [+o anthm] by ChanServ |
13:50.27 | cjk | hi, how good is the asterisk spool for .call files. what happes when i put 10k files in it |
13:50.59 | Ariel_ | 10k wow that is allot. Depends on your box. But that just might over load it. |
13:51.04 | tzafrir | cjk, I guess they'll be processed in turn. |
13:51.15 | [TK]D-Fender | cjk : Use the "mv" method to shift them in and I guess it should work.... just a question of concurrency which I couldn't say.... |
13:51.19 | docelmo | Usually there is something going on but its just dead today |
13:51.33 | [TK]D-Fender | docelm0 : Why do you think I'm here :) |
13:51.35 | Assid | oh.. i thought if a call was done via sip uri straight to the 2nd box.. would be different |
13:51.36 | tzafrir | Chances are asterisk will not handle easily 10000 concurrent calls. |
13:51.48 | Assid | my bad |
13:52.24 | *** join/#asterisk Lino` (n=Lino@i577BCDCA.versanet.de) |
13:52.36 | cjk | ok, 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-Fender | Assid : 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-Fender | Assid : make sure both calls pass through THAT PHONE'S local server. |
13:57.01 | Assid | but 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 |
13:58.12 | file | meep? |
13:59.24 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
13:59.32 | *** join/#asterisk jaike (i=jaike@124.106.139.4) |
13:59.36 | *** join/#asterisk jero (n=jero@savoirfairelinux.net) |
14:00.56 | jaike | quick question guys. anyone experiencing crashes using mixmonitor? |
14:02.00 | wunderkin | jaike, what version are you using |
14:02.09 | jaike | 1.2.7.1 |
14:02.24 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
14:02.24 | jaike | its not very often..maybe once every 1000 calls |
14:03.19 | wunderkin | theres nothing in bugs, get a bt and post it |
14:03.47 | Ecio | is there any call manager expert here? im tryin to strip a digit for calls between CM and Asterisk but i have no success |
14:03.59 | wunderkin | unless you try the 1.2 branch, i know there have been some fixes but dont know if anything past 1.2.7.1 |
14:04.22 | jaike | our 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 |
14:05.09 | wunderkin | update first, make sure it is producing core files, if still happens then report |
14:05.41 | jaike | hmmm..ok will do that..thanks |
14:07.14 | MrChimpy | hey guys |
14:07.54 | MrChimpy | i 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 :) |
14:09.03 | Assid | MrChimpy: ask the question if anyone can help.. they just will |
14:09.50 | MrChimpy | well, 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? |
14:12.19 | sevard | riddle me this |
14:12.21 | brettnem | hey anyone in here using the SIP jitterbuffer? |
14:12.42 | sevard | you have a four line phone yet a line now and then goes UNREACHABLE |
14:12.56 | brettnem | MrChimpy: If you put more than one of those cards into the same box you will need to have unique ID. |
14:13.21 | brettnem | MrChimpy: BTW, buying equipment from Digium has nothing to do with anyone's willingness to answer a question. ;) |
14:13.45 | brettnem | sevard: what kind of phone is this? |
14:14.02 | sevard | Aastra 480i CT |
14:14.23 | sevard | it happens on Sipura 2002 ATAs too, one line will go unreachable for 20 or some odd minutes |
14:15.06 | blitzrage | brettnem: 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 |
14:15.25 | file | blitzrage: we are really trying to get it in |
14:15.34 | file | it's just... Russell and I tried yesterday, and the trunk version is toast |
14:15.46 | file | seems like everyone has been testing the 1.2 version |
14:15.49 | brettnem | blitzrage: 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 |
14:15.52 | MrChimpy | brettnem: thanks :) |
14:16.13 | brettnem | sevard: that is weird. maybe a nat issue? Do they all use the same port? |
14:16.31 | sevard | different ports behind a switch NAT'd once |
14:16.32 | brettnem | sevard: btw, I was just trying to get my first aastra 480i CT up yesterday coincidentially |
14:16.44 | sevard | brettnem: they're great phones |
14:16.48 | brettnem | sevard: might be a port translation issue (broken router) |
14:16.55 | sevard | do you want an example cfg file for your tftp server? |
14:17.15 | brettnem | sevard: 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 |
14:17.28 | brettnem | <dialtone> <digit> <silence><digit><dialtone> |
14:17.29 | sevard | that... weird |
14:17.37 | brettnem | yeah |
14:17.39 | sevard | check your firmware version, i'll check against mine |
14:17.56 | brettnem | ok, it's a customer's phone, I'll ask when they get in. |
14:18.12 | eset | hi, 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 |
14:18.19 | sevard | Application: Version 1.3.0.1080 SIP Boot ROM: Version: 1.1.0.4 |
14:18.35 | brettnem | sevard: I'll check |
14:18.42 | sevard | brettnem: alright, if you need config files to provision it hit me a line when they come in |
14:18.42 | blitzrage | file: yah - I read something about that yesterday... thats really good bad :( |
14:18.51 | sevard | I 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) |
14:19.11 | brettnem | sevard: looks neat, but I'm not sure what advantage it has over a nice cordless phone and a sipura |
14:19.30 | sevard | brettnem: 9 lines, four cordless RF phones |
14:19.37 | brettnem | sevard: that is nice |
14:19.39 | sevard | brettnem: big buttons, nice display |
14:19.52 | sevard | brettnem: it's WAY easier to xfer and conf and intercom on this phone than regular phones |
14:19.58 | brettnem | sevard: I haven't actually seen it yet.. my customer is like 600 miles away ;) |
14:20.05 | sevard | brettnem: services, directory listing, awesome quality intercom |
14:20.15 | brettnem | cool |
14:20.20 | sevard | brettnem: :) i love that phone, my favorite part though is the weight in the handset |
14:20.36 | sevard | brettnem: just the perfect amount |
14:20.36 | Assid | hey [TK]D-Fender, any update on the 1.6.6 when polycom plans to release it for us 'regular' folks? |
14:21.01 | brimstone | eset, did you get your 2.6.8-2 problem fixed yet? |
14:21.28 | eset | brimstone : no |
14:21.45 | brettnem | bbiab |
14:21.56 | *** join/#asterisk sb_mx (n=sb_mx@200.94.154.226) |
14:22.29 | brimstone | eset, set EXTRAVERSION to = "-2" in /lib/modules/`uname -r`/build/Makefile |
14:22.36 | brimstone | then recompile zaptel |
14:22.44 | eset | ah |
14:23.43 | file | brimstone knows too much, hit him! |
14:24.08 | brimstone | <zoidberg> look! i'm helping! |
14:24.11 | eset | brimstone : thanks, i will see how this goes and let you know :) |
14:24.23 | brimstone | okey dokey eset! |
14:28.00 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@195.167.202.197) |
14:31.53 | eset | its official, brimstone does know too much :) |
14:32.06 | brimstone | oh noes! |
14:32.37 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
14:32.48 | eset | (thanks) |
14:33.08 | brimstone | you're welcome eset |
14:33.25 | brimstone | does 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) |
14:36.59 | Henk | I'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 ? |
14:37.41 | jake1932 | manager should be running on that port |
14:37.54 | jake1932 | is manager enabled in manager.conf? |
14:38.59 | jake1932 | btw - agi is different from manager |
14:39.31 | *** part/#asterisk buzzyd (n=buzzyd@82-45-247-173.cable.ubr01.enfi.blueyonder.co.uk) |
14:39.51 | Henk | jake1932, 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 |
14:40.44 | *** join/#asterisk gcarrillog (n=gcarrill@201.152.19.192) |
14:43.42 | *** join/#asterisk jaike (i=jaike@210.5.119.146) |
14:43.47 | Henk | jake1932, 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) |
14:44.19 | mut | sweeeeeeet |
14:44.19 | mut | http://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) |
14:44.33 | Henk | the 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) |
14:45.45 | Supaplex | if only it gave you cash back |
14:45.54 | Skarmeth | hi all |
14:46.25 | blitzrage | heh... you can't find girls that skinny in the USA anymore |
14:47.56 | jake1932 | Henk: you can drop a .call file when the request comes from the web |
14:48.21 | Ecio | is there a tool/command in CLI that shows what dialplan is applied when u try to call a number? |
14:49.15 | Ecio | something like "what happens when i dial 4666 from here?" |
14:49.20 | blitzrage | Ecio: 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...) |
14:49.39 | blitzrage | Ecio: no -- what happens depends on what you have in the dialplan |
14:49.42 | Skarmeth | When 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 : |
14:49.42 | Skarmeth | 2 } ) |
14:49.43 | jake1932 | Ecio: set verbose 4 |
14:49.46 | Ecio | uhm |
14:50.07 | jake1932 | Ecio: or higher |
14:50.24 | blitzrage | Skarmeth: ${EXTEN:length:offset} is the format |
14:50.59 | blitzrage | Skarmeth: to keep the first two digits, you do: ${EXTEN::2} |
14:51.07 | Skarmeth | blitzrage, like ${EXTEN:2:1} ? |
14:51.13 | blitzrage | Skarmeth: sure |
14:51.13 | Skarmeth | ok |
14:51.35 | SpaceBass | this disco video is AWESOME! |
14:51.49 | blitzrage | EXTEN=ABCDEFG -> ${EXTEN:2:1} -> BC |
14:52.05 | blitzrage | to go from the other end... use negative numbers |
14:52.06 | Skarmeth | blitzrage, when it is done, ${EXTEN} value continues unchanged right? it just a copy of it... |
14:52.09 | mut | :) |
14:52.16 | blitzrage | Skarmeth: yes, ${EXTEN} is unchanged |
14:52.16 | Henk | jake1932, 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) |
14:52.19 | Skarmeth | blitzrage, thanks |
14:52.54 | blitzrage | that disco video sucked |
14:52.55 | jake1932 | Henk: nope - you can use the .call file to point to a point in your dialplan to process the call |
14:53.10 | blitzrage | jake1932: you use the Local/ channel for that |
14:53.36 | jake1932 | blitzrage: he wants to call it from a web app |
14:53.40 | blitzrage | ahhh |
14:54.22 | blitzrage | http://www.youtube.com/watch?v=dMH0bHeiRNg |
14:57.08 | Henk | jake1932, 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. |
14:57.42 | Henk | combined with ajax the user will be completely baffeled |
14:58.08 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
14:58.44 | jake1932 | Henk: there's plenty of cool stuff to do - just seems for your app, dropping a simple .call file would work just fine |
15:00.31 | Henk | jake1932, 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 |
15:01.03 | jake1932 | Henk: we already do that (the same way) |
15:01.29 | Henk | nice... what company ? |
15:01.39 | jake1932 | Henk: the .call file just start the call - you'll throw the meat in the dialplan |
15:01.46 | Delta239 | how can i access the home directory on asterisk? |
15:01.55 | jake1932 | Henk: a big cable company :) |
15:02.07 | eset | okeokde, now i get a saddening "Invalid module format" for ztdummy when i modprobe it on 2.6.8 |
15:02.15 | Delta239 | the only thing i have here is 4 files with no folders |
15:03.13 | Henk | jake1932, 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) |
15:04.11 | jake1932 | Henk: 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) |
15:04.24 | jake1932 | totally different application |
15:05.00 | Henk | do you do this with php? java? .. |
15:05.05 | jake1932 | .net |
15:05.51 | jake1932 | we made a custom app to interface directly through port 5038 |
15:06.28 | *** join/#asterisk tiwyant (n=twyant@pix.wyantcomputerservices.com) |
15:06.48 | tiwyant | Mornin' |
15:07.44 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
15:08.00 | asterboy | good morning...good morning...good morning the whole day through... |
15:08.07 | tiwyant | that too |
15:08.28 | tiwyant | Anyone around know if there's a limit on how many mailboxes I can include in a group voicemail? |
15:08.42 | asterboy | I took some VIagra, CIaliz, and something else that gave me 500% cum volume...good morning! |
15:08.47 | Henk | jake1932, 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 |
15:09.05 | jake1932 | Henk: correct |
15:09.59 | jake1932 | asterboy: i just got that same e-mail this morning |
15:10.03 | jake1932 | asterboy: and yesterday |
15:10.11 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
15:10.19 | asterboy | lol...I get them every 5 min |
15:10.54 | mercestes | Why would I want 500% cum volume? Wait.....nevermind....don't tell me. I dont' wanna know. |
15:11.08 | Henk | jake1932, 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? |
15:11.11 | asterboy | drown her in your pleasure! |
15:11.12 | MrChimpy | dunno. girls get thirsty |
15:11.51 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
15:11.54 | jake1932 | Henk: are you exposing asterisk on the net? |
15:12.08 | mercestes | meh.....I didn't wanna know, I said. |
15:12.11 | MrChimpy | weird. i've installed 2nd TE411P, it works and I can dial to it, but zap status shows Alarms on the working port as NOP |
15:12.32 | jake1932 | Henk: IOW - i have a mythtv setup with a bunch of ports open that's completly safe |
15:12.40 | blebleble | anyone 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 |
15:13.45 | jake1932 | Henk: it's not accessible to the public internet though |
15:13.45 | asterboy | Anyone here experienced with paging systems? |
15:13.46 | Henk | jake1932, for incomming calls yes. I want to just redirect calls to one of our 24-hour support staff members' cellphones after office hours. |
15:13.46 | eset | anyone know why the svn of zaptel gives a "module format invalid" when you try to modprobe the zaptel modules? |
15:13.46 | tiwyant | what kind of paging? |
15:13.47 | asterboy | simple, handset to paging horn. |
15:13.48 | noky | i 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 |
15:13.52 | noky | any idea????? |
15:14.17 | *** join/#asterisk prog (n=prog@vdsoft.kh-net.cz) |
15:14.21 | tiwyant | asterboy: Find yourself a nice fxo/fxs capable paging amp and do it that way. Works like a charm. |
15:14.25 | prog | hello to all |
15:14.47 | tiwyant | blebleble: Have you edited the op_buttons.cfg and changed the sizes? |
15:14.53 | asterboy | ya, I have the LUPCM Bogen |
15:15.12 | asterboy | Do you use an FXO or FXS to hookup the paging system? |
15:15.12 | jake1932 | Henk: who are you protecting the system from, callers? |
15:15.21 | asterboy | I'm thinking FXO |
15:15.28 | prog | if someone says: "our SIP provider doesn`t require registering" - what does it mean ? How can I solve this in sip.conf ? ( register=> IP_address ) ? |
15:15.29 | blebleble | tiwyant: i have not, i see the defaults which ones should i be changing? |
15:15.31 | tiwyant | Hmm |
15:15.46 | jake1932 | Henk: website visitors? |
15:15.48 | tiwyant | asterboy: I think FXO is how I set the last one up. |
15:16.06 | asterboy | ya, otherwise the paging system is going to need to figure out what to do when called. |
15:16.10 | tiwyant | blebleble: lemme check, real quick |
15:16.32 | blebleble | tiwyant: thanks there just is a lot of crazy options, rectangle etc etc |
15:16.38 | tiwyant | asterboy: The last one I set up that way just answered the call and let me talk. |
15:16.54 | jake1932 | Henk: re: phpagi, i'm not familiar with the security issues |
15:16.59 | asterboy | that was fxs then if it answered. |
15:17.37 | asterboy | or 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) |
15:18.42 | tiwyant | blebleble: 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. |
15:19.08 | tiwyant | asterboy: I'm pretty sure I had an fxo port into it because I was using the other port for backup 911 |
15:19.16 | tiwyant | but the device was switchable |
15:19.21 | asterboy | ah |
15:19.26 | asterboy | I think mine is also. |
15:19.36 | asterboy | The other solutino is to go out the sound card |
15:19.42 | tiwyant | yuck |
15:19.52 | tiwyant | I'd rather go fxo/fxs and not screw with the sound scard |
15:19.54 | asterboy | but then, I'll need an amp and run speaker wire |
15:19.57 | tiwyant | yeah |
15:20.00 | *** join/#asterisk slobberknocker (n=ckwall@63.149.122.94) |
15:20.05 | asterboy | ya, I agree |
15:20.13 | asterboy | more to go wrong at the * side |
15:21.22 | asterboy | tiwyant, what did you have for the speakers? |
15:21.31 | tiwyant | quick 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 |
15:21.46 | tiwyant | asterboy: In ceiling whatever jobbies they had with the Merlin system I replaced |
15:21.58 | asterboy | Mine either wants an amp or 70V speakers, which I'm guessing need to be powered |
15:22.12 | tiwyant | yes |
15:22.31 | asterboy | Sure like to get a unit that has its own amp. |
15:22.40 | tiwyant | I agree, that's the way to go |
15:22.44 | Henk | jake1932, 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 |
15:23.26 | jake1932 | Henk:write a tight dialplan and any callers or callees won't be able to do anything |
15:24.05 | asterboy | This 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) |
15:24.25 | slobberknocker | ok, 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. |
15:24.26 | Henk | jake1932, 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 |
15:24.44 | jake1932 | Henk: good luck with it |
15:24.44 | blitzrage | slobberknocker: it doesn't work like that |
15:24.51 | asterboy | ah forget paging...I'm going to start my own CLEC: http://www.voip-info.org/wiki/view/How+to+start+a+Clec |
15:24.51 | slobberknocker | ok |
15:25.11 | tiwyant | CLEC = lots of paperwork and crap |
15:25.18 | tiwyant | I'm just going to start a phone sex line |
15:25.19 | jake1932 | [TK]Defender lol |
15:25.25 | tiwyant | using voip over the internet to india |
15:25.34 | blebleble | tiwyant: 755 |
15:25.46 | tiwyant | thanks blebleble |
15:25.54 | asterboy | ok, forget clec...I hagte paperwork |
15:26.00 | slobberknocker | well 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. |
15:26.13 | asterboy | back to 500% cum volume sales |
15:26.19 | slobberknocker | i get the error, exten 6406 not found in context progrexion. |
15:26.24 | slobberknocker | it is under evolution |
15:26.32 | slobberknocker | so how do i specify? |
15:27.05 | Delta239 | anybody here knows about astguiclient? |
15:27.24 | tiwyant | In your extensions.conf do you have an exten => entry for 6406 in the progrexion context? |
15:27.24 | tiwyant | Cause the error you posted says you don't |
15:27.26 | jake1932 | slobberknocker: [progrexion] doesn't have an extension 6406 |
15:27.46 | slobberknocker | right... it is under a different context. |
15:27.48 | jake1932 | you need to modify your extensions.conf file |
15:27.56 | slobberknocker | but i have to specify a context in zapata, right? |
15:27.59 | [TK]D-Fender | jake1932 : Whats so funny? |
15:28.04 | slobberknocker | well i will have multiple contexts. |
15:28.13 | slobberknocker | how do i make the zapa.conf use all of them? |
15:28.49 | jake1932 | nick changing |
15:28.56 | jake1932 | it is already |
15:29.12 | [TK]D-Fender | slobberknocker : 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) |
15:29.35 | jake1932 | slobberknocker: you just need to include exten => 6406,1,Something in your progrexion context |
15:30.11 | blebleble | tiwyant: hay for the scale and margin is that on the icon, led, or arrow ? |
15:30.16 | slobberknocker | ok, 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? |
15:30.54 | jake1932 | slobberknocker: yes |
15:31.06 | slobberknocker | ok, great. Thanks for the help |
15:31.14 | jake1932 | thank 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) |
15:32.11 | jake1932 | [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-Fender | jake1932 : Originated from myplaying Tribes 1 CTF. I tended to "watch the fort", hence the nick. |
15:33.56 | MrChimpy | frikkin weird. even zttool says NOP for the single span I'm actually using yet it still works |
15:34.21 | jake1932 | [TK]D-Fender: ok |
15:34.34 | [TK]D-Fender | slobberknocker : Ok, maybe we should start a little further back. What kind of technologies are you using with your server? |
15:35.06 | [TK]D-Fender | jake1932 : [TK] was my old Action:Half-Life clan. |
15:35.10 | slobberknocker | are you meaning am i using a T1, etc? |
15:35.17 | [TK]D-Fender | slobberknocker : yes |
15:35.29 | slobberknocker | i have a TE410P with 2 pris |
15:35.42 | [TK]D-Fender | slobberknocker : Describ every kind of device taking calls into your system, VoIP pproviders, T1/E1/Analog, everything |
15:36.13 | slobberknocker | all providers are T1 and all devices are polycom 301 and 501s |
15:36.54 | [TK]D-Fender | slobberknocker : 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-Fender | slobberknocker : Excellent phone choices. |
15:38.22 | slobberknocker | ok, 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] |
15:38.22 | slobberknocker | exten => _1XXXNXXXXXX,1,Dial(Zap/g2/${EXTEN:1}) |
15:38.22 | slobberknocker | exten => _1800NXXXXXX,1,Dial(Zap/g2/${EXTEN:1}) |
15:38.22 | slobberknocker | exten => _NXXXXXX,1,Dial(Zap/g2/801${EXTEN:0}) |
15:38.34 | slobberknocker | and on the inbound i will do all of my dids and ivrs. |
15:38.58 | slobberknocker | then on those dont i have to do something to the effect of include => {other contexts} |
15:39.11 | [TK]D-Fender | slobberknocker : 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? |
15:39.16 | slobberknocker | the main goal is for tenanting. i need to have different companies on one system |
15:39.28 | [TK]D-Fender | slobberknocker : Since I now know you're on PRI your dialplan needs a MASSIVE overhaul. |
15:39.35 | [TK]D-Fender | slobberknocker : PM |
15:39.39 | slobberknocker | ok |
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) |
15:44.38 | Ecio | guys 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. |
15:44.52 | Ecio | I can call from the CM to the asterisk but not viceversa, i got a "Dial failed due to CHANUNAVAIL" in the debug |
15:45.11 | Ecio | the strange thing is that i see this in the debug of the not working one: |
15:45.11 | _Paulo_ | ~a@h |
15:45.12 | Ecio | Dial("SIP/60666-170e", "/4666|120|W |
15:45.20 | kaz0358 | quick 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-Fender | Ecio : that dial statement is VERY wrong... |
15:45.45 | Ecio | yeèp |
15:45.48 | Ecio | the working one is |
15:45.51 | Ecio | Dial("SIP/70002-2658", "SIP/CCM1/4666") |
15:46.05 | noky | any expert of realtime extensions of asterisk ???? |
15:46.05 | [TK]D-Fender | Ecio : And I doubt anyone here is going to want to hear about your systems running A@H. Please read the channel topic. |
15:46.08 | Ecio | i cant understand what's happening.. i've just copied the macro.. :) |
15:46.40 | xy_goat | hi all - how can i stop jitter from happening on my asterisk install? |
15:46.51 | Ecio | d-fender: actually im fed up of this a@h... most of the times when u do something in the web interface everything's messed... |
15:47.15 | kaz0358 | ecio, the configuration files aren't that hard to manage.. |
15:47.15 | Ecio | u 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 |
15:48.34 | Ecio | i see |
15:50.02 | Ecio | but on asterisk.org there are only sources, not binaries, isnt it? |
15:50.16 | kaz0358 | would 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-Fender | Ecio : 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) |
15:50.58 | CunningPike | [TK]D-Fender: Tell us how you really feel ;) |
15:51.09 | Ecio | d-fender :) |
15:52.13 | [TK]D-Fender | Should have checked your magazine! |
15:52.21 | blitzrage | booo |
15:52.21 | [TK]D-Fender | :D |
15:52.24 | blitzrage | roll playing sucks :) |
15:52.30 | blitzrage | role* even |
15:53.04 | _Paulo_ | Ecio, most linux distros have * packaged |
15:53.13 | [TK]D-Fender | Actually I'm just waiting for the new forge models to arrive before I prepare to buy my next one... |
15:53.24 | Ecio | uhm, 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" |
15:53.33 | Katty | [TK]D-Fender: beep! |
15:53.35 | eset | hmmm |
15:53.35 | [TK]D-Fender | Ecio : And for the love of God avoid those packages LIKE THE PLAGUE |
15:53.43 | eset | i would think about that twice (debian and asterisk) |
15:53.44 | [TK]D-Fender | Katty : ! ! ! |
15:53.45 | Ecio | d-fender which packages ? a@h ? :D |
15:53.45 | [TK]D-Fender | ;) |
15:53.49 | blitzrage | ugh-- don't use packages... learn how to build Asterisk -- and it's not even hard |
15:53.52 | Katty | [TK]D-Fender: you set off my hilight. |
15:53.57 | *** join/#asterisk tamp4x (n=Lab@64.201.13.172) |
15:54.09 | blitzrage | and I hate debian, so what do I know? :) |
15:54.09 | sevard | :( |
15:54.12 | sevard | i don't get any hugs anymore |
15:54.13 | tamp4x | does 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... |
15:54.17 | eset | but good luck if you need zaptel |
15:54.47 | [TK]D-Fender | blitzrage : You hate Debian? Very respectable distro. Just avaoid with a very few particular pckages and its gold... |
15:54.47 | eset | debian is giving me monstrous trouble wiht module formats for zaptel, eek |
15:54.57 | sevard | Does anyone have any experience with DirectVNC? |
15:55.32 | myiagy | i have quite a few systems running debian and asterisk.. with 2.4 and 2.6 kernels.. including on a AMD64.. never had much trouble |
15:55.56 | blitzrage | [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 |
15:55.57 | eset | no trouble with building the zaptel drivers on 2.6? |
15:55.58 | Ecio | eset: 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. |
15:56.23 | eset | Ecio : you need zaptel for the conferences even with sip |
15:56.25 | Ecio | blitz: if debian is a bitch, what's gentoo? :D |
15:56.27 | Katty | [TK]D-Fender: pfft. |
15:56.28 | myiagy | eset hm, no.. it build just fine.. |
15:56.47 | Ecio | eset: dont they use "virtual" modules? |
15:57.10 | [TK]D-Fender | Katty : ;) |
15:57.10 | eset | ztdummy is a virtual module for zaptel as far as i can tell, wont modprobe with 2.6 |
15:57.18 | Ecio | lol |
15:57.32 | blitzrage | Ecio: a whote -- I hate gentoo too |
15:57.39 | blitzrage | -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. |
15:57.49 | file | I like Debian. |
15:57.55 | blitzrage | gentoo is for people who have far too much time |
15:58.05 | Ecio | blitz :) |
15:58.06 | [TK]D-Fender | Gentoo is for ricers! |
15:58.08 | eset | it seems there are a few peole though having the same problem |
15:58.17 | eset | looks like it is somewhat historical |
15:58.24 | tamp4x | why would i get a 404 not foudn when i recv and options message |
15:58.29 | asterboy | no, lfs is for people with too much time. |
15:58.39 | eset | 2.6 gives a "invalis module format" error for zaptel modules |
15:58.51 | Katty | eset: :< |
15:59.01 | sevard | i love you |
15:59.01 | sevard | fag |
15:59.04 | Katty | [TK]D-Fender: i think i'm about to put my very first production box up at a client's. |
15:59.15 | mercestes | Gentoo is cruel and unusual. |
15:59.16 | Katty | [TK]D-Fender: i'm all skeered inside :< |
15:59.18 | myiagy | eset i got that error once. but i think the problem was with gcc version |
15:59.23 | mercestes | I love you too, Sevard... |
15:59.23 | mercestes | homo. |
15:59.26 | eset | with gcc, really? |
15:59.33 | asterboy | brokeback *....again! |
15:59.41 | mercestes | Gentoo is text based chinese water torture. |
15:59.41 | myiagy | kernel was compiled with gcc 4.. and i compiled zaptel with gcc 3 |
15:59.45 | eset | you mean a diff gcc version from the one used to compile the kernel maybe |
15:59.48 | eset | ah, yes |
15:59.53 | eset | ok, i will look at that |
15:59.56 | Ecio | eset: so what distro do u suggest |
16:00.02 | asterboy | LFS! |
16:00.03 | eset | um.... |
16:00.07 | eset | atari? |
16:00.09 | eset | ;) |
16:00.11 | asterboy | lol |
16:00.13 | Ecio | aster: i've not enought time... |
16:00.15 | Ecio | -t |
16:00.26 | Ecio | but maybe on UZIX on MSX |
16:00.34 | Ecio | it could handle 1 sip call in half duplex mode |
16:00.37 | eset | im going for paper cups and string |
16:00.37 | Ecio | =) |
16:01.01 | eset | most reliable telephoney platform there ever was |
16:01.25 | mercestes | I advocate kororaa.org |
16:01.28 | Ecio | eset: uhm and being practical? :D |
16:01.30 | mercestes | I run it. |
16:01.30 | [TK]D-Fender | Katty : Doing this for a business now? Not just intenal consumtion? |
16:01.59 | mercestes | it's gentoo based without the torture, crying and suicide attempts. |
16:02.06 | Katty | [TK]D-Fender: yes'm :> |
16:02.08 | eset | Ecio : if debian works with thise gcc tip then i'll let you know |
16:02.13 | Katty | [TK]D-Fender: :> :< :> :<, etc. |
16:02.53 | [TK]D-Fender | Katty: Your personal consulting on on behalf of your company (as I'm not really sure where you work) |
16:03.47 | Katty | [TK]D-Fender: it's a leetle place. |
16:03.50 | eset | seems my kernel was compiled with the same version of gcc, so dont think tahts it |
16:04.10 | file | bbl |
16:04.18 | Katty | kbi |
16:04.30 | *** part/#asterisk slobberknocker (n=ckwall@63.149.122.94) |
16:04.55 | [TK]D-Fender | Katty : So was that "you consulting for you" or "you doing it for your job"? |
16:04.55 | myiagy | eset i think the gcc error says something about "version magic" so yours might not be the same error i got |
16:06.11 | eset | ok, hmmm |
16:06.24 | eset | it was a good idea though |
16:06.44 | myiagy | i had the invalid module thing too.. just can't remember what it was.. i'll tell you if i remember.. |
16:07.29 | myiagy | eset do you have kernel-headers installed? |
16:07.33 | eset | yep |
16:07.48 | myiagy | what version of kernel you running |
16:07.55 | eset | 2.6.8-2 |
16:08.51 | *** join/#asterisk slobberknocker (n=ckwall@63.149.122.94) |
16:09.16 | eset | myiagy : tried with module-assistant, as well as svn, and the latest tar |
16:09.28 | eset | and apt-get of course |
16:09.32 | *** join/#asterisk vechers (n=103326C9@64.61.117.138) |
16:09.37 | Katty | [TK]D-Fender: oh. |
16:09.41 | Katty | [TK]D-Fender: it's for a company, not me. |
16:09.51 | Katty | [TK]D-Fender: though i'm really the only one here that knows anything about asterisk. |
16:09.57 | Katty | [TK]D-Fender: and, admittedly, that ain't a whole lot at all :< |
16:10.59 | eset | maybe i just try a diff gcc anyway |
16:11.07 | myiagy | eset can you copy the whole error? |
16:11.45 | eset | i use this: modprobe ztdummy, and i get this error: |
16:11.55 | eset | WARNING: Error inserting zaptel (/lib/modules/2.6.8-2-686/zaptel/zaptel.ko): Invalid module format |
16:12.08 | eset | FATAL: Error inserting ztdummy (/lib/modules/2.6.8-2-686/zaptel/ztdummy.ko): Invalid module format |
16:12.13 | eset | FATAL: Error running install command for ztdummy |
16:12.19 | RaYmAn-Bx | eset: check dmesg |
16:12.41 | eset | zaptel: no version for "struct_module" found: kernel tainted. |
16:12.47 | eset | 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' |
16:12.58 | RaYmAn-Bx | it's that last one |
16:13.00 | *** join/#asterisk Coyotee (n=root@sipx.ica.net) |
16:13.01 | eset | yeah |
16:13.03 | eset | hmmm |
16:13.16 | mercestes | kernel tainted. I like that error. |
16:13.16 | *** join/#asterisk wiseguy_ (n=chivilis@infospalvos.lt) |
16:13.21 | wiseguy_ | hellow |
16:13.25 | Coyotee | hey all |
16:13.29 | Hymie | I'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? |
16:13.43 | eset | so how do i work around that? |
16:13.55 | Hymie | eset: you can use modprobe -f to force it to insert anyhow |
16:14.13 | wiseguy_ | how do i write extension if i want to execute system command after call? |
16:14.21 | eset | -f doesnt work, same error |
16:14.26 | Hymie | eset: er |
16:15.00 | RaYmAn-Bx | eset: using a distribution kernel or self-compiled? |
16:15.15 | Coyotee | does 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) |
16:15.27 | myiagy | eset the headers you have installed |
16:15.30 | eset | dist kernel |
16:15.34 | myiagy | are not the same for the kernel you have running |
16:15.46 | eset | hmmm |
16:15.52 | myiagy | i 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 |
16:15.54 | RaYmAn-Bx | myiagy: they shouldn't be either (supposedly) |
16:16.20 | myiagy | they shouldn't? |
16:16.30 | Holos | Anyone 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. |
16:16.32 | myiagy | don't you need smp headers if you have an smp kernel |
16:16.46 | RaYmAn-Bx | that might be true |
16:16.48 | eset | the version is (from /proc/version : Linux version 2.6.8-2-68 |
16:17.03 | wiseguy_ | help me, someone |
16:17.06 | eset | and the headers are kernel-headers-2.6.8-2 |
16:17.29 | myiagy | try updating the headers then |
16:17.32 | eset | ok |
16:18.21 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
16:18.27 | mut | on a dark desert highway |
16:18.29 | prog | 21:14:24.162830 IP 192.168.0.250.5060 > 113.151.64.101.5060: UDP, length: 359 |
16:18.29 | prog | 21:14:24.176512 IP 113.151.64.101.51782 > 192.168.0.250.5060: UDP, length: 358 |
16:18.30 | mut | cool wind in my hair |
16:18.40 | mut | warm smell of something, rising up through the air |
16:19.05 | prog | why opposite site ( 113.xxxxx ) replies with 51782 port ? |
16:19.33 | prog | this can`t work |
16:20.04 | mercestes | mut: when off in the distance..I saw a shimmering light. |
16:20.17 | mut | so i called up the captain, please bring me my wine |
16:20.19 | mercestes | mut: my head grew heavy and my sight grew dim...I had to stop for the night. |
16:20.32 | mut | :P |
16:20.39 | mut | everyone together now! |
16:20.45 | [TK]D-Fender | mercestes : Oh the two of you can stop ANY TIME now... |
16:20.50 | mut | liviin it up at the hotel california! |
16:20.58 | mut | heh |
16:21.07 | mut | man i wish today would end already |
16:21.12 | mut | hour and a half and i'm gone |
16:21.42 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
16:21.48 | eset | well, it cured the first error |
16:21.58 | wiseguy_ | ghem |
16:22.01 | *** part/#asterisk Holos (n=asdf@204.101.26.106) |
16:22.02 | wiseguy_ | someone, help me |
16:23.05 | eset | myiagy : it works! |
16:23.07 | wiseguy_ | i have a problem writing extension. I wan't to execute System command after end of all calls in that context? |
16:23.31 | eset | if i knew who u were and i had any money you'd be in my will ;) |
16:23.42 | myiagy | eset ;) |
16:23.46 | eset | thans |
16:23.54 | eset | thanks, that saved me a real headache |
16:24.57 | eset | now i'm gonna document it to save someone else a headache |
16:25.01 | sevard | Does _anyone_ have either fbvnc or DirectVNC working |
16:25.09 | mercestes | Hey, Wiseguy. Guess what I found under google "asterisk extensions execute system command. |
16:25.11 | mercestes | This link |
16:25.12 | mercestes | http://www.voip-info.org/wiki/index.php?page=Asterisk+-+documentation+of+application+commands |
16:25.16 | sevard | haha |
16:25.36 | mercestes | May the Google be with you, young padawan. |
16:25.48 | mercestes | Ironically, the command you are looking for is called System. |
16:26.13 | mercestes | Check out this link too. http://www.catb.org/~esr/faqs/smart-questions.html |
16:26.24 | slobberknocker | shouldnt it be that when i have one phone in one context, they should not be able to dial a phone from another context? |
16:27.01 | wiseguy_ | mercestes: okay, show me the answer, because i haven't found there something special |
16:27.10 | mercestes | ............... |
16:27.20 | mercestes | well.....the little letters...... |
16:27.21 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
16:27.25 | mercestes | if you slur them all together, make words. |
16:27.46 | mercestes | and 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) |
16:30.10 | flujan | ping coppice |
16:30.21 | wiseguy_ | mercestes: i not asking how to execute the command, i'm asking how to write exact extension for all ended calls |
16:30.23 | mercestes | slobberknocker: 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. |
16:30.38 | wiseguy_ | mercestes: i mean s, 103, o what? |
16:30.40 | flujan | guys, i want to configure asterisk to work with a legacy proprietary pbx system. |
16:30.57 | flujan | First of all, I plugged the E1 interfaces using a crossover T1 cable. |
16:30.58 | mercestes | slobberknocker: The phone itself is arbitrary in that statement. |
16:31.11 | flujan | then, I compile asterisk to work with the mfc r2 signalling. |
16:31.27 | mercestes | wiseguy_: specifically after answered calls? |
16:31.49 | mercestes | wiseguy_: Or after all calls period regardless of handling? |
16:31.54 | flujan | The digium card's led just flash a red light... should it be green? |
16:32.13 | wiseguy_ | mercestes: no, after all clauses, i want the "fact" about ended call.. |
16:32.21 | flujan | anyone here already see a woriking E1 card? |
16:33.03 | jake1932 | i had one working |
16:33.23 | jake1932 | you can use zttool to see if you have any alarms |
16:33.35 | flujan | jake1932, have it a green light ? |
16:33.47 | flujan | jake1932, yes, zttool give no errors. |
16:34.00 | flujan | jake1932, how do you have your zaptel.conf? |
16:34.16 | jake1932 | i'm not running it anymore in e1 mode - using t1 now |
16:34.50 | jake1932 | when running zttool, and you get a call, are you noticing any bit flips? |
16:35.04 | mercestes | wiseguy_: Could do phpAGI, and have it hang until the call is done and deliver the handling code to you. |
16:35.56 | mercestes | wiseguy_: or you could do a Goto(s-${DIALSTATUS},1) |
16:35.56 | mercestes | <PROTECTED> |
16:36.32 | mercestes | wiseguy_: But if I recall correctly, a Answered/Hangup() scenario will terminate at the hangup() and not continue priority jumping. |
16:36.45 | flujan | jake1932, http://pastebin.com/735315 |
16:36.57 | flujan | i got the above message when I try to make a call. |
16:37.45 | JunK-Y | zap show status |
16:39.34 | mercestes | wiseguy_: I think using AGI is going to be the only real way to handle it for an answer situation, but I could be mistaken. |
16:39.53 | flujan | JunK-Y, cat /proc/zapte/1 |
16:40.09 | flujan | JunK-Y, can I type this instead? |
16:40.46 | Hymie | does anyone know why a caller would just sit in a queue, and never timeout? |
16:40.53 | flujan | JunK-Y, someone here already configure a E1 card? |
16:40.56 | JunK-Y | flujan: actually isnt the same output. |
16:41.16 | JunK-Y | not me, since im in North America. |
16:41.42 | flujan | JunK-Y, http://pastebin.com/735335 |
16:42.35 | flujan | JunK-Y, A E1 link has 30 channells... When I configure it, the configurations spans to a second channell. |
16:42.36 | jake1932 | <PROTECTED> |
16:42.38 | JunK-Y | so ur span2 is all r ight. |
16:43.06 | flujan | jake1932, sorry, I'm only using the cat /proc/zaptel/1 |
16:43.32 | flujan | JunK-Y, I have just one E1 link. the span 2 is not connect to the other computer. |
16:43.47 | flujan | TE406P |
16:43.51 | flujan | I have this card |
16:44.06 | Hmmhesays | JunK-Y |
16:44.14 | JunK-Y | hey Hmmhesays ! |
16:44.22 | flujan | jake1932, how can I debug about the alarm? |
16:44.30 | Hmmhesays | been awhile, how are you? |
16:44.39 | *** join/#asterisk salviadud (n=ralfalfa@dsl-201-129-72-124.prod-infinitum.com.mx) |
16:44.41 | JunK-Y | im fine, summer is comin', ya? |
16:44.44 | jake1932 | flujan: start with a loopback connector |
16:44.45 | *** join/#asterisk BugKham (i=CKGLOB@125.24.7.45) |
16:44.54 | Hmmhesays | about damn time it si |
16:44.55 | Hmmhesays | *is |
16:45.06 | Hmmhesays | you're coming to cluecon aren't you? |
16:45.13 | JunK-Y | probably yes. |
16:45.15 | JunK-Y | u? |
16:45.16 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
16:45.53 | BugKham | any news about 1.2.8? |
16:45.54 | Hmmhesays | yeah I think so |
16:45.56 | flujan | jake1932, just to confirme |
16:46.04 | flujan | jake1932, to confirm... |
16:46.08 | flujan | jake1932, http://www.jaredsmith.net/misc/cables/ |
16:46.42 | flujan | jake1932, I used a straight connector and a T1 crossover is it right? |
16:46.44 | salviadud | has anyone here been to Defcon? |
16:46.47 | jake1932 | http://kb.digium.com/entry/1/95/ |
16:47.34 | *** part/#asterisk BugKham (i=CKGLOB@125.24.7.45) |
16:48.01 | flujan | jake1932, thanks |
16:48.17 | flujan | jake1932, I just want to make sure about the crossover T1. Is that correct? |
16:48.48 | jake1932 | crossover cable != loopback connector |
16:49.37 | flujan | jake1932, and to configure a E1 interface: http://pastebin.com/735358 |
16:50.09 | jake1932 | flujan: did you make a loopback connector according to http://kb.digium.com/entry/1/95/? |
16:50.20 | flujan | jake1932, not yet |
16:50.53 | flujan | jake1932, Must I start from a straight t1 cable and apply that changes? |
16:51.00 | jake1932 | it's good to have that for troubleshooting |
16:51.22 | jake1932 | flujan: you can butcher a straight t1 cable |
16:51.35 | jake1932 | just make sure you have a spare :) |
16:55.13 | Hymie | does anyone know why a caller would just sit in a queue, and never timeout? |
16:55.31 | Hymie | it 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) |
16:57.06 | u168138 | evening all, i`m trying to connect two asterisk boxs using sip, has anyone done it before? |
16:57.36 | sevard | wow, 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) |
16:58.59 | Delta239 | how do i modify a command like to edit a file |
16:59.29 | *** join/#asterisk masonf (n=masonf@dungle.vineyard.net) |
17:00.05 | AndyC | anybody got 2 asterisk boxes connected together using SIP before? |
17:01.02 | Hmmhesays | sevard stfu and gbtw |
17:01.38 | masonf | is it possible keep the pstn ringing after a call gets to a zatpel card? |
17:01.39 | sevard | Hmmhesays: lickamaballsa |
17:03.05 | Hmmhesays | i was an 1 1/2 late for work this morning |
17:03.15 | C4T3l | great scott! |
17:03.20 | jake1932 | slacker |
17:03.29 | Hmmhesays | <shrug> they don't pay me enough |
17:04.00 | mercestes | How does that make you unique?? |
17:04.59 | sevard | upi |
17:05.08 | sevard | you were one and a half late for work? |
17:05.36 | *** part/#asterisk masonf (n=masonf@dungle.vineyard.net) |
17:06.22 | sevard | Delta239: uhh, what |
17:06.57 | Delta239 | what is the command to modify a file |
17:07.07 | sevard | use whatever editor you want |
17:07.13 | sevard | vi, vim, emacs, pico, nano |
17:07.17 | Delta239 | and how do i exit from the editor |
17:07.21 | Delta239 | ok i used vi |
17:07.24 | sevard | what fucking editor are you using |
17:07.25 | mercestes | I suggest vi. |
17:07.28 | sevard | :q! |
17:07.32 | mercestes | It's the easiest. |
17:07.34 | sevard | I don't suggest vi |
17:07.38 | sevard | pico is easier. |
17:07.43 | Delta239 | how do i exit vi |
17:07.45 | sevard | wayyyyyyyyyyyy easier |
17:07.47 | mercestes | Vi is easier. |
17:07.54 | jake1932 | Delta239: look up |
17:07.55 | sevard | Delta239: press escape and type q! |
17:07.58 | [TK]D-Fender | Tastes great! Less filling! |
17:08.16 | sevard | mercestes: dude, the learning curve for pico/nano is like 2.2 seconds, for vi you litterally have to look up commands to learn it |
17:08.24 | Delta239 | and does it automatically saves |
17:08.31 | mercestes | sevard: Not if you know what the commands mean in it's mother language of klingon. |
17:08.36 | sevard | Delta239: use wq to save and exit |
17:08.50 | mercestes | sevard: Which only makes sense since VI is clearly the klingon word for "Edit." HELLO!?! |
17:08.50 | sevard | mercestes: f0r teh ubar n3rd |
17:09.05 | Delta239 | thanks |
17:11.59 | *** join/#asterisk vijatit (n=vijay@61.11.90.90) |
17:14.33 | vijatit | Can 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) |
17:16.35 | sevard | Does 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-Fender | Polycom IP 430 appears to have displaces the IP 501 as my first-run business phone choice :) |
17:21.18 | vijatit | gsm |
17:21.27 | *** join/#asterisk lorinc (n=ang@caracas-4553.adsl.interware.hu) |
17:21.42 | [TK]D-Fender | Likely a category killer.... |
17:22.11 | _Paulo_ | vijatit, change to a codec with better compression is not an option? |
17:22.37 | vijatit | I have used the "packetization=" directive for sip clients and have made calls at low bw |
17:23.18 | vijatit | actually, 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-Fender | vijatit : You are TRUNKING that IXA link I hope.... |
17:24.27 | Delta239 | sevard: 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 |
17:24.31 | Delta239 | it wont exit vi |
17:24.36 | *** join/#asterisk jayk- (i=jayk@lasziv.reprehensible.net) |
17:24.38 | vijatit | even with g729, the required bw is 8kbps for data and 32kbps for ip/udp/rtp overheads, as there are frequent packets. |
17:24.50 | jayk- | is there a way to put the asterisk configuration into a pgsql database and have it read from that? |
17:24.56 | mercestes | lol |
17:25.14 | mercestes | Delta239: Vi does not coddle the weak! Here is what you do.. |
17:25.21 | *** join/#asterisk ToTo (n=ToTo@host135-167.pool872.interbusiness.it) |
17:25.37 | mercestes | Delta239: Tap esc twice. press : (That's ; while holding the shift key) |
17:25.46 | mercestes | Delta239: Enter wq and press enter. |
17:25.46 | sevard | Delta239: press escape and type :wq |
17:25.57 | vijatit | but, 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-Fender | vijatit : Are you TUNKING your IAX calls or not? |
17:26.02 | Delta239 | ahhhhhhhhhhhhhhhhh |
17:26.04 | Delta239 | :D |
17:26.09 | Delta239 | xD |
17:26.14 | mercestes | Delta239: See, isn't klingon easy? |
17:26.15 | vijatit | that is the sole idea of packetization directive in sip |
17:26.21 | Delta239 | yeah hehe |
17:26.25 | Delta239 | thanks |
17:27.00 | vijatit | D-Fender, actually, this is for calls from clients. These calls come as one call per dest and hence cant be trunked |
17:27.10 | mercestes | Delta239: I would say your welcome but klingon has no such expression. So, english wil have to suffice. |
17:27.23 | sevard | Delta239: 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. |
17:28.05 | mercestes | sevard: ......it took me longer than 3 seconds........*cries* That's just hurtful. |
17:28.31 | jake1932 | "Any retard who spends 3 seconds" |
17:28.43 | jake1932 | lo |
17:28.46 | jake1932 | l |
17:28.48 | _Paulo_ | Delta239, vi has a very steep learning curve, but it pays after years... :-) |
17:28.55 | [TK]D-Fender | vijatit : 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. |
17:29.12 | mercestes | Here ya go. http://www.thinkgeek.com/homeoffice/mugs/7bbe/ |
17:29.26 | [TK]D-Fender | vijatit : So moving to 40ms packets chops your overhead in half |
17:29.50 | vijatit | D-Fender : ya. thats the point i was making |
17:30.03 | vijatit | now, is there a way to do the same with iax? |
17:30.54 | *** join/#asterisk diclophis (n=diclophi@65.203.37.58) |
17:30.58 | diclophis | hello all |
17:31.12 | vijatit | D-Fender : With sip, at this stage, there is a patch available and after applying htat, its just a single configuration addition in sip.conf |
17:31.14 | diclophis | lets say i have 2 PRI 'log terms' plugged into two seperate asterisk boxes |
17:31.20 | diclophis | is it possible to dial one from the other? |
17:31.20 | [TK]D-Fender | vijatit : 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. |
17:32.38 | Delta239 | whatever dude i thought i could ask you guys but thanks anyway... next time i won't ask this types of questions |
17:33.20 | diclophis | _Paulo_ yea i have both boxes setup to accept calls for their respective terminated numbers |
17:33.23 | mercestes | :( Now see what you did, Sevard...you hurt his feelings. |
17:33.28 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
17:33.30 | diclophis | however when i try dialing from one machine to the other, it wont connect |
17:33.55 | _Paulo_ | how are you dialing? |
17:34.05 | diclophis | and i get a bunch of "Don't know what to do with control frame 15" messages |
17:34.16 | diclophis | just a standard dial originate through a manager api |
17:34.30 | vijatit | The 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. |
17:34.50 | diclophis | and i get a "PROGRESS with cause code 31 received" |
17:34.57 | diclophis | thats on the dialing machine |
17:34.58 | _Paulo_ | vijatit, what is on the client side? |
17:35.05 | diclophis | the answering machine never registers any call |
17:35.17 | vijatit | an iax softphone |
17:35.49 | _Paulo_ | vijatit, your softphone supports speeks? |
17:36.23 | vijatit | both ends are under our control, both interms of code and configs. Currently, it doesnt have speex, but i could add it |
17:36.33 | vijatit | will adding speex help? |
17:38.27 | _Paulo_ | vijatit, I think it pays just try it out. |
17:39.26 | vijatit | i'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 |
17:41.22 | flujan | jake1932, the loopback interface works |
17:41.40 | flujan | jake1932, I have two green leds |
17:42.15 | diclophis | would it make any difference that the 2 machines are in the same building? |
17:42.19 | jake1932 | flujan: great! |
17:42.24 | flujan | jake1932, http://pastebin.com/735449 |
17:42.26 | *** join/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
17:42.40 | jake1932 | flujan: much better |
17:42.41 | flujan | jake1932, why appears the span 3 |
17:42.51 | flujan | jake1932, since I just have two spans configured |
17:42.55 | Peaceful | can you set values from the telnet interface on a Cisco 7960 phone? |
17:42.55 | Nugget | telnet is eeeeeeevil! |
17:43.07 | jake1932 | flujan: 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) |
17:45.29 | flujan | jake1932, http://pastebin.com/735457 |
17:45.37 | flujan | jake1932, just two... |
17:45.53 | flujan | jake1932, this is a strange behavior, isn't it? |
17:46.11 | jake1932 | indeed |
17:46.42 | diclophis | here 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" |
17:46.49 | jake1932 | flujan: someone else may have more insight on why it is happening |
17:47.07 | sevard | OH MY GOD IT FRIGGEN WORKS |
17:47.21 | *** part/#asterisk slobberknocker (n=ckwall@63.149.122.94) |
17:47.22 | flujan | guys, i configure just two e1 links in my asterisk box... in the zap show status appear three configured |
17:47.33 | flujan | someone can help me with that? |
17:49.03 | flujan | jake1932, 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) |
17:50.23 | gbodemantv | hi |
17:50.41 | gbodemantv | is anyone using multiple servers and one central VM location |
17:51.24 | diclophis | as in hosting one voicemail extension on a central server that is connected to other servers through iAX? |
17:52.29 | gbodemantv | I have 3 servers, each with its own voicemail |
17:52.56 | gbodemantv | problem is that the voicemail on one does not see to want to talk to the others |
17:52.57 | diclophis | oh, but the voicemail dir is hosted on NFS? |
17:53.05 | gbodemantv | it was |
17:53.16 | gbodemantv | tried that |
17:53.18 | diclophis | yea, the app_voicemail with asterisk sucks |
17:53.31 | gbodemantv | but it started serious lag and problems |
17:53.37 | gbodemantv | had to rollback to local voicemail |
17:53.42 | diclophis | are you using realtime mysql config extension? |
17:53.47 | gbodemantv | yes |
17:54.02 | diclophis | what is the problem? |
17:54.47 | gbodemantv | with 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 |
17:54.53 | gbodemantv | then just hangs up |
17:55.04 | gbodemantv | the NFS box is across fiber to another office though |
17:55.11 | gbodemantv | I think that is the issue |
17:55.34 | _Paulo_ | gbodemantv, use something like rsync |
17:55.50 | jake1932 | <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 |
17:56.41 | Peaceful | Anyone know the cisco syntax to set a value from the telnet interface on a 7960? (assuming you can do such a thing) |
17:57.00 | flujan | jake1932, yes, i will first try to understand why I configure 2 and appears 3 configured E1 link |
17:57.04 | mercestes | Peaceful: Try links. |
17:57.35 | mercestes | Peaceful: 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. |
17:58.01 | gbodemantv | is it better just to have all servers sent to my main server for voicemail |
17:58.03 | bzbw | hi, anyone know if i can use this: exten => _6XXX,hint,SIP/${EXTEN} ? show hint gives me empty result |
17:58.16 | Flauto | hi people, how does senddtmf work? |
17:58.21 | diclophis | gbodemantv, if i were using the default asterisk voicemail i would have it hosted off one machine |
17:58.27 | Flauto | like if i dial a phone number with ivr |
17:58.29 | Strom_C | bzbw: you need a priority |
17:58.36 | diclophis | mainly because of how asterisk stores the voicemail files |
17:58.46 | Flauto | wait for a few seconds and senddemf with the extension number? |
17:58.49 | diclophis | you could be running into NFS locks or something |
17:58.50 | gbodemantv | and just have the other 2 servers refer to it in extensions.conf? |
17:58.59 | bzbw | Strom_C: So if I give it a priority, it will work? |
17:59.00 | diclophis | yea, through a IAX connection |
17:59.09 | Strom_C | bzbw: supposedly |
17:59.11 | diclophis | if i am not mistaken with IAX you can make it seem transperent |
17:59.23 | Strom_C | you need to get your extensions language syntax correct :) |
17:59.36 | mercestes | STrom_C: hints do not have a priority. |
17:59.43 | mercestes | strom_C: That I am aware of. |
17:59.48 | Strom_C | *shrug* |
17:59.52 | diclophis | like somehow merge the dialplans from the boxes |
18:00.13 | mercestes | bzbw: I would try entering in a few "hints" without the wildcards and see if that works for you first. |
18:00.14 | bzbw | Strom_C: thx. |
18:00.24 | Strom_C | I'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) |
18:00.57 | mercestes | Strom_C: Weren't you that guy that was mean to me earlier? |
18:01.09 | mercestes | yea..I remember you. |
18:02.30 | flujan | jake1932, how can I check if my car TE406P is configured as E1's and not T1's ? |
18:03.05 | jake1932 | flujan: ztcfg -vv |
18:03.19 | jake1932 | flujan: should show the correct number of channels |
18:03.24 | jake1932 | flujan: might be other ways |
18:03.44 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
18:04.17 | jake1932 | flujan: or zttool |
18:05.22 | flujan | jake1932, ztcfg -vv show the 62 channels... |
18:05.46 | jake1932 | flujan: any errors? |
18:05.53 | flujan | jake1932, no... |
18:06.04 | jake1932 | flujan: good |
18:06.06 | flujan | jake1932, but and about the output of the zap show status? |
18:06.37 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
18:06.53 | jake1932 | flujan: ? |
18:07.19 | flujan | jake1932, zap show status shows me three configured spans... but I have only two... :P |
18:07.25 | jake1932 | :) |
18:07.43 | paolob | Hi 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) ? |
18:07.57 | jake1932 | flujan: already told you i don't know - answer still remains |
18:08.16 | flujan | jake1932, ok... thanks... :D |
18:08.25 | flujan | I think I will try the list |
18:09.00 | Peaceful | mercestes: does the 7960 have a web interface? It doesn't seem to respond on port 80. |
18:09.22 | mercestes | Peaceful: yes...it does. |
18:09.53 | *** join/#asterisk Hymie (i=hymie@L8R.net) |
18:10.02 | Hymie | does anyone know why calls never timeout in my queue? |
18:10.07 | Hymie | the default of 5 minutes, or any other time, it doesn't matter. I can leave a call in the queue for hours even |
18:10.50 | jake1932 | Peaceful: it has a telnet interface |
18:11.01 | jake1932 | Peaceful: different port |
18:11.38 | jake1932 | Peaceful: actually, a telnet server, and a minibrowser client |
18:11.46 | jake1932 | Peaceful: no web server |
18:12.37 | Peaceful | jake1932: sooo, I should try browsing to the telnet port? |
18:14.59 | Peaceful | hmm..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. |
18:15.44 | Peaceful | Back 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. |
18:16.14 | Strom_C | Peaceful: why do you need to set values via the telnet interface in the first place? |
18:17.20 | Dr-Linux | we are going to buy a new server for our TE210P cards, brand should be DELL or Sun .. |
18:17.26 | Dr-Linux | any recommendations? |
18:17.48 | gmfm | Hymie: when you use the Queue app, option 'n' will prevent it from retrying on timeout and go to the next step in the dialplan |
18:17.49 | Peaceful | Because 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 |
18:18.26 | bzbw | Strom_C: tried with priority when using _6xxx for hints, it does not work. |
18:18.46 | Strom_C | Peaceful: that's why you test with a phone locally and get it working /before/ you deploy the thing :) |
18:18.53 | diclophis | why would my sip register attempts be timing out? |
18:19.05 | Peaceful | Dr-Linux: Opterons |
18:19.21 | Dr-Linux | Peaceful: what's Opterons? |
18:19.49 | Dr-Linux | Peaceful: this server will be running high IVR solutions |
18:19.52 | Peaceful | Strom_C: And how do I test the home-user's NAT setup here??? I've got * ->NAT-> Internet ->NAT-> home user cisco |
18:20.23 | Strom_C | Peaceful: if you're doing SIP, you should never have more than one NAT in the connection. SIP does not like to play with NAT. |
18:20.44 | Peaceful | Dr-Linux: Dell's all use Intel processors for now, Sun sells really expensive AMD opterons. I was just recommending AMD Opterons over Intel anything. |
18:20.45 | Strom_C | having the telephone and the asterisk box behind NAT is a recipe for disaster |
18:20.58 | Peaceful | Strom_C: So I've noticed. |
18:20.59 | Strom_C | or, more accurately, behind different NATs |
18:21.13 | Peaceful | Strom_C: Signalling works fine, I'm just not getting voice |
18:21.32 | Dr-Linux | Peaceful: can you suggest me any from Sun then? |
18:21.37 | Strom_C | Peaceful: yes, thats the problem |
18:21.42 | Strom_C | it's a NAT issue |
18:21.43 | vijatit | i had asked this question an hour back, since the channel is active now, i think i'll ask it again to try my luck |
18:21.46 | Peaceful | Strom_C: which is annoying, since they both seem to be UDP ports |
18:21.54 | Strom_C | no amount of twiddling with the phone will get it working |
18:22.02 | Strom_C | fix your broken network setup |
18:22.07 | Peaceful | Dr-Linux: If you've got the money, yes. |
18:22.33 | Hmmhesays | the tick tock of the clock is painful, all sane and logical, I want to tear it off the wall |
18:22.37 | Dr-Linux | Peaceful: what about X4100 from Sun? |
18:23.09 | vijatit | has anyone tried iax with custom packetization duration. (In sip, this is possible with "packetization=" directive ). |
18:23.16 | Strom_C | Hmmhesays: 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 :) |
18:23.53 | Peaceful | Dr-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. |
18:24.06 | Dr-Linux | anybody likes DELL servers? :) |
18:24.12 | Delta239 | me |
18:24.21 | Delta239 | :D |
18:24.23 | *** join/#asterisk mtaht4 (n=m@reserve-64-79-114-30.wiline.com) |
18:24.31 | Strom_C | Dr-Linux: I have a TE4xxP working in a Dell Poweredge server |
18:24.31 | salviadud | dell rocks da house yo |
18:24.45 | Dr-Linux | Delta239: you like DELL over others? |
18:24.51 | Delta239 | i don't have one but i worked for Dell for years |
18:24.54 | Dr-Linux | Strom_C: what's the server model? |
18:25.06 | Strom_C | Poweredge something-or-other :) |
18:25.17 | Dr-Linux | guys recommend me any DELL server |
18:25.17 | Strom_C | dont remember |
18:25.21 | Strom_C | it's at a client's prem |
18:25.24 | *** join/#asterisk nagl (n=nagl@86.59.54.237) |
18:25.27 | Dr-Linux | Strom_C: i need model number |
18:25.31 | Delta239 | the poweredge |
18:25.37 | Hmmhesays | and boobies |
18:25.42 | Hmmhesays | apparently there is a shortage in pakistan |
18:25.46 | Strom_C | yes, cant forget the boobies |
18:26.12 | Dr-Linux | we have more then 70 DELL poweredge servers and 30 Sun |
18:26.28 | Dr-Linux | but we need buy a new one for Asterisk/TE210P |
18:26.32 | Strom_C | I think Hmmhesays was talking about a shortage of boobies |
18:26.46 | Dr-Linux | so need recommended models |
18:27.03 | Dr-Linux | Hmmhesays: how many boobies? :) |
18:27.17 | protocoldoug | i just installed the zaptel drivers (to use ztdummy) but when i issue "zap show channels" * says: "no such command zap", any ideas? |
18:27.18 | Hmmhesays | preferably an even number |
18:27.39 | Dr-Linux | lolz |
18:27.59 | Dr-Linux | Hmmhesays: 32 bits boobies? :P |
18:28.03 | _Paulo_ | protocoldoug, try "load chan_zap.so" |
18:28.06 | Hmmhesays | yeah 31 is bad, parity error |
18:28.35 | Dr-Linux | Hmmhesays: 36 also errors but, it's cool :P |
18:28.58 | protocoldoug | _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) |
18:29.13 | Dr-Linux | hhm.. so no recommendation? :S |
18:29.20 | _Paulo_ | protocoldoug, you are welcome |
18:29.27 | wasabi | There 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. |
18:33.14 | wasabi | Why do you say that? |
18:38.04 | _Paulo_ | wasabi, because I like to program for fun. |
18:38.07 | Dr-Linux | http://www.dell.com/downloads/global/products/pedge/en/2850_specs.pdf |
18:38.12 | wasabi | k... |
18:38.12 | salviadud | im gooooooing home, bye dudes |
18:38.14 | Dr-Linux | hows this server? |
18:40.03 | *** join/#asterisk gursikh (n=FreePBX9@adsl-68-92-36-133.dsl.hstntx.swbell.net) |
18:40.25 | bzbw | looks like i can't use "_6XXX, hint, SIP/${EXTEN}", this is painful for 100 extensions:( |
18:40.37 | Ariel_ | 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 |
18:45.40 | Hmmhesays | shaking that ass on the floor |
18:45.41 | *** join/#asterisk dapatrick (n=dapatric@static-151-204-184-67.pskn.east.verizon.net) |
18:45.46 | Hmmhesays | bumpin and grinding that pole |
18:46.03 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
18:46.08 | Strom_C | taaaake ooooon meeeeeeee |
18:48.26 | *** join/#asterisk TonyM (n=TonyM@softins.claranet.co.uk) |
18:48.26 | Hymie | gmfm: 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) |
18:54.31 | sevard | Strom_C: Take me onnn |
18:54.43 | sevard | TAKKKEEEE MEEEEE ONNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNNN |
18:55.00 | Strom_C | I'll bee gooooone |
18:55.12 | Strom_C | hahaha |
18:55.36 | Hmmhesays | anyone know what the name of that new korn song is? |
18:55.38 | sevard | it's scientifically proven to get that song stuck in your head for weeks at a shot |
18:55.46 | Strom_C | wow, this is timing |
18:55.50 | Strom_C | they just called |
18:56.22 | Hmmhesays | ahyone? |
18:56.24 | Hmmhesays | *anyone |
18:56.51 | sevard | Strom_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-Fender | bzbw : not that bad... |
18:58.40 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
19:02.53 | Strom_C | sevard: 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) |
19:04.34 | sevard | Strom_C: but not a ..general test number? |
19:06.50 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
19:07.45 | bzbw | D-Fender: why, I hate to manage those long list of extensions. |
19:08.13 | Hymie | does anyone know why calls never timeout in my queue? |
19:08.17 | Hymie | the 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) |
19:09.29 | sandra78 | help!!! |
19:09.30 | *** join/#asterisk lorinc (n=ang@caracas-4553.adsl.interware.hu) |
19:09.42 | sandra78 | :S |
19:09.48 | _Paulo_ | what is your problem, sandra78? |
19:10.09 | sevard | wow |
19:10.12 | sevard | this guy on the phone |
19:10.18 | sevard | just tried to teach me how to use wget |
19:10.26 | sevard | seriously. |
19:11.01 | Peaceful | so...he's like "type w-g-e-t", or what? |
19:11.15 | *** join/#asterisk lorinc (n=ang@caracas-4553.adsl.interware.hu) |
19:11.18 | sandra78 | <_Paulo_> Does anybody knows how to change the asterisk ringing song? i want to record my own ringing song |
19:11.21 | sevard | people 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 |
19:11.43 | sevard | Peaceful: 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? |
19:12.01 | sandra78 | no, i mean the false ringing |
19:12.06 | sevard | sandra78: do you mean music on hold or indication |
19:12.46 | sandra78 | when you call an extension you can hear a ring |
19:12.47 | sevard | indication. |
19:12.47 | sevard | look at the wiki |
19:12.57 | sandra78 | i want to chenge this song for another song similar to pstn song |
19:13.03 | Hymie | hum.. is anyone here even using queues? |
19:13.14 | sevard | Hymie: I am |
19:13.29 | [TK]D-Fender | bzbw :its a 1-shot job mostly.... pretty quick to make... |
19:13.40 | _Paulo_ | sandra, are you in Brazil? |
19:13.47 | Hymie | FOR THE LOVE OF GOD MAN, any ideas? ;) |
19:13.56 | sevard | what's the porblem? |
19:14.03 | Hymie | my queues never timeout |
19:14.11 | Hymie | people sit in them for 2000 years, if they please |
19:14.12 | [TK]D-Fender | Hymie : Paste the line you use to call the queue |
19:14.13 | sevard | whats your syntax |
19:14.14 | brad_mssw | sevard: 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 |
19:14.37 | sevard | brad_mssw: no, i'm flipping bits and shit, i talk more professional on the phone :) |
19:14.42 | Hymie | exten => s,2,Queue(tech-queue|t||30) |
19:14.54 | Hymie | 30 is a test, I've tried realistic numbers, like 300, etc |
19:15.19 | _Paulo_ | sandra78, are you in Brazil? |
19:15.32 | Dr-Linux | guys, 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) |
19:15.44 | sevard | Hymie: my queue is exten => s,4,Queue(co_queue|tT|||300) |
19:15.55 | Hymie | sevard: it just never, ever times out here |
19:16.02 | Hymie | sevard: what version of asterisk? |
19:16.03 | Dr-Linux | Three total: three PCI-X® slots (64-bit/133MHz) or two PCI Express™ |
19:16.03 | Dr-Linux | slots (1 x 4 lane and 1 x 8 lane) and one PCI-X slot (64-bit/100MHz) |
19:16.06 | sevard | Hymie: try my syntax |
19:16.13 | [TK]D-Fender | Hymie : You are missing a "|" |
19:16.17 | sevard | 1.2.7.1 |
19:16.30 | Hymie | hmm.. ok, I will try this, damned ||| intead of || |
19:16.30 | [TK]D-Fender | exten => s,2,Queue(tech-queue|t|||30) <- this is right |
19:16.47 | [TK]D-Fender | Hymie : though you'll want to change that 30.... |
19:16.56 | Hymie | [TK]D-Fender: sure.. I want 600 anyhow |
19:16.56 | sevard | Hymie: most problems are silly syntax problems |
19:16.59 | Dr-Linux | put another T to make your life more easy :) |
19:17.03 | Hymie | [TK]D-Fender: just had to try everything |
19:17.06 | [TK]D-Fender | Hymie : You should have been ...RTFM! |
19:17.19 | Hymie | why another T? I don't want the caller to transfer anywhere, except to one digit extensions |
19:17.19 | [TK]D-Fender | Hymie : "show application queue" |
19:17.40 | Dr-Linux | anybody get a change to look into the link that i pasted? |
19:17.43 | sevard | I believe the T is queue out, but I don't recall |
19:17.46 | [TK]D-Fender | Hymie : I didn't add the "t" I assumed you WANTED it there so I LEFT it in from your line. |
19:18.10 | sevard | [TK]D-Fender: you LOVE to do THIS a LOT |
19:18.18 | [TK]D-Fender | Hymie : "tT" are worthless if you're using real phones... |
19:18.20 | Hymie | [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 |
19:18.25 | sevard | [TK]D-Fender: at least you don't do this |
19:18.35 | [TK]D-Fender | sevard : It compensates for lack of volume control :) |
19:18.41 | sevard | [TK]D-Fender: :| |
19:19.18 | [TK]D-Fender | Hymie : What kind of phones are you using? |
19:19.25 | Hymie | Uniden 200's |
19:19.29 | Hymie | quite nice actually |
19:19.30 | sevard | [TK]D-Fender: worthless? refresh my memory. |
19:19.32 | Hymie | I like them |
19:19.38 | Hymie | although they always clip at the start of the call :/ |
19:19.41 | Hymie | even with latest firmware |
19:19.44 | [TK]D-Fender | Hymie : My condolences... get rid of the "t" you don't need it, they have SIP transfer capability. |
19:20.04 | Hymie | no big or small t? we can't have any 't'? but I like t ;) |
19:20.04 | [TK]D-Fender | Hymie : I regret having bougth the 2 that I did.... |
19:20.09 | Hymie | I have a poster of Mr. T on the wall |
19:20.13 | Hymie | [TK]D-Fender: why? |
19:20.14 | sevard | mmm tea |
19:20.14 | [TK]D-Fender | Hymie : Niether. remove it completely. |
19:20.22 | Hymie | [TK]D-Fender: what don't you like? |
19:20.24 | [TK]D-Fender | Hymie : Why what? My purchase regrets? |
19:20.28 | Hymie | yeah |
19:20.41 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
19:20.47 | Hymie | ah, jolly good, thanks for the | assist, it's fine now |
19:20.53 | [TK]D-Fender | Hymie : Feels friggen cheap, shitty interface, inflexible provisioning, poor button placement, etc |
19:20.54 | Hymie | half a day wasted on that |
19:20.58 | sevard | I regret buying a PAP2 thinking I might be able to hax it |
19:21.15 | Hymie | [TK]D-Fender: sure, they feel cheap, but they ARE cheap too ;) |
19:21.17 | [TK]D-Fender | Hymie : Polycom IP 301 is worth every extra penny. |
19:21.22 | sevard | Hymie: google, site:voip-info.org <keyword>, saves time. |
19:21.26 | Hymie | [TK]D-Fender: well, tell that to my client ;) |
19:21.30 | Hmmhesays | i've never played with the 301's |
19:21.37 | Hymie | sevard: ? I've been there, I just didn't notice the extra | missing |
19:21.50 | Hymie | sevard: one of those things... it just happens sometimes |
19:21.51 | sevard | glasses? |
19:21.57 | sevard | dyslexia? |
19:22.00 | [TK]D-Fender | Hymie : 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. |
19:22.00 | sevard | i'm dyslexic as fuck |
19:22.03 | Hymie | sevard: ah, I'll make sure to hand you a Mr. Perfect star later ;) |
19:22.14 | Hymie | sevard: I just missed it, I usually don't |
19:22.23 | sevard | why does this remind me of that movie |
19:22.27 | Hymie | [TK]D-Fender: cool |
19:22.34 | flujan | Hi all, I'm having problems. Every time I try to place a call using asterisk I receive: http://pastebin.com/735639 |
19:22.40 | sevard | some guy says some dude told his crazy sister there were ghosts in her clothes |
19:22.41 | [TK]D-Fender | Hmmhesays : The 301 is pretty nice, I use one at home as well as a 501. I use 60x's at work mostly. |
19:22.42 | sevard | damn. |
19:22.52 | [TK]D-Fender | Hmmhesays : the new IP 430 is going to severly ROCK.... |
19:22.53 | Hmmhesays | I have a snom 190 at home |
19:22.53 | Hymie | hehe |
19:23.16 | [TK]D-Fender | Hmmhesays : A category killer I'm sure.... |
19:23.26 | Hymie | anyhow, thanks guys.. off to finish this build |
19:23.28 | Hmmhesays | say what? |
19:23.42 | [TK]D-Fender | Hmmhesays : The new Polycom IP 430.... |
19:23.43 | Hmmhesays | whats up with the 430? looks like a 501 |
19:24.00 | [TK]D-Fender | Hmmhesays : nope, its built on a 301 fram, but with MASSIVE improvements |
19:24.08 | [TK]D-Fender | frame* |
19:24.09 | Hmmhesays | such as |
19:24.14 | sevard | oh man, this jerky doesn't taste hot |
19:24.17 | sevard | but when it gets in your stomach |
19:24.21 | sevard | you regret eating it |
19:24.27 | *** part/#asterisk Hymie (i=hymie@L8R.net) |
19:24.43 | tdonahue-laptop | thanks sevard, just a little too much information though... |
19:24.48 | [TK]D-Fender | Hmmhesays : Full pixel LCD, 4 soft keyes, arrorw key navigation, PoE INTEGRATED, LED's for the line keys, speakerphone and more. |
19:24.57 | sandra78 | <_Paulo_> Hi you mean do i have to change in the indication.conf file? |
19:25.00 | sevard | tdonahue-laptop: i do NOT look forward to that coming out of my brown star |
19:25.01 | [TK]D-Fender | Hmmhesays : 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... |
19:25.27 | Hmmhesays | whats the price on these? |
19:25.40 | Hmmhesays | yeah i was reading that [TK]D-Fender |
19:25.46 | sevard | those phones look like crap |
19:25.55 | _Paulo_ | sandra78, you can change the tones there. |
19:25.57 | [TK]D-Fender | Hmmhesays : Its slated to fit betweent he 301 & 501, so it may push each a little or just push the 301 down I suspect. |
19:26.04 | sevard | the Aastra 480i already has that AND more |
19:26.27 | *** join/#asterisk nialp (n=nialp@217-162-135-208.dclient.hispeed.ch) |
19:26.30 | sevard | wayyyyyyyyyyyyyyy more |
19:26.33 | Hmmhesays | i have yet to find a better speakerphone than polycom |
19:26.36 | [TK]D-Fender | Hmmhesays : 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 |
19:26.46 | sevard | aastra :) |
19:26.50 | Hmmhesays | whats the speakerphone like on teh 301? |
19:27.11 | [TK]D-Fender | Hmmhesays : NONEXISTANT :) |
19:27.22 | [TK]D-Fender | Hmmhesays : but he SPEAKER is just like the others :) |
19:27.24 | Hmmhesays | oh... that out for any of my installs then |
19:27.36 | nialp | after nearly 1 year of service my asterisk with passiv HFC-S in NT mode doesn't supply a dial tone any more |
19:27.51 | sandra78 | exten => 264,3,Playtones(!950/330,!1400/330,!1800/330,0) |
19:27.52 | nialp | i upgraded to asterisk 1.2 |
19:28.02 | nialp | and still no dial tone |
19:28.27 | Hmmhesays | that suggests hardware failure |
19:28.27 | nialp | i 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-Fender | Hmmhesays : 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 |
19:28.58 | Hmmhesays | when is it coming out? |
19:29.03 | [TK]D-Fender | Hmmhesays : About a month |
19:29.15 | [TK]D-Fender | Hmmhesays: I am SO gonna pawn off my IP 301 for it :) |
19:29.24 | Hmmhesays | i'll get a contract to buy me one |
19:29.27 | [TK]D-Fender | Hmmhesays : with any luck AT PAR :) |
19:29.43 | Qwell[] | [TK]D-Fender: got a link? |
19:29.48 | Hmmhesays | [TK]D-Fender have you ever played with openwrt? |
19:29.49 | [TK]D-Fender | Qwell : see above |
19:30.00 | sandra78 | i was using exten => 264,3,r |
19:30.04 | Qwell[] | gotcha |
19:30.18 | *** join/#asterisk vooduhal (n=vooduhal@tc-proxy2.catt.com) |
19:30.25 | [TK]D-Fender | Hmmhesays : Nope, I have the best grade router to choose for it but never got around to playing with it. |
19:30.33 | Qwell[] | I'll wait for the 630 |
19:30.40 | vooduhal | Hey guys. Can anyone point me in the direction of a MIB for the Polycom IP phones? |
19:30.47 | Hmmhesays | i have an audiocodes ac494 board sitting here |
19:30.50 | Hmmhesays | running linux |
19:30.55 | [TK]D-Fender | Qwell : 630? Fictitious? Or linkable? |
19:30.57 | Hmmhesays | fxs and fxo ports |
19:31.06 | Qwell[] | [TK]D-Fender: fictitious |
19:31.17 | Qwell[] | this has no mini browser |
19:31.19 | Damin | How much is the 430 going to be selling for retail? |
19:31.19 | Qwell[] | lame! |
19:31.21 | [TK]D-Fender | Qwell : thats for gettin my hopes up! |
19:31.30 | Qwell[] | Damin: I don't think it's been stated |
19:31.38 | [TK]D-Fender | Damin : As I'm told, in between the 301 & 501 |
19:31.42 | Qwell[] | [TK]D-Fender: What would you guess though, $140? |
19:31.52 | Flauto | May 24 14:31:09 WARNING[17213]: app_dial.c:1162 dial_exec_full: Invalid timeout specified: '+asterisk' |
19:31.55 | [TK]D-Fender | Qwell : roughly in my estimate. |
19:31.58 | Flauto | what is that |
19:32.12 | Damin | [TK]D-Fender: My scrollback is fucked.. |
19:32.16 | Qwell[] | 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. |
19:32.26 | Qwell[] | Damin: ^ |
19:32.44 | Qwell[] | [TK]D-Fender: heh, I didn't even see that line...I guessed |
19:32.53 | [TK]D-Fender | For full PoE + Brick, and all the other plusses it looks kinda sick. |
19:33.10 | [TK]D-Fender | Qwell : Both fair guess' :) |
19:33.18 | flujan | http://pastebin.com/735639 someone have a idea why this is happening? |
19:33.39 | flujan | i'm in trying this in my loopback environment. |
19:34.36 | JackEStorm | Damin: look it up on froogle |
19:34.50 | Qwell[] | JackEStorm: it helps if the product existed |
19:34.59 | vooduhal | Also, does anyone know of decent mib viewer for *nix? |
19:36.25 | [TK]D-Fender | vooduhal : What is this about MIB? |
19:36.56 | vooduhal | Just needing to know if there a polycom specific MIB for the IP phones, specifically 500 and 600s. |
19:37.05 | [TK]D-Fender | Damin, Hmmhesays, Qwell : Also integral to the release of the 430 is SIP 2.0 firmware. <- |
19:37.17 | Qwell[] | 2.0, pfft |
19:37.22 | Qwell[] | cisco is already on 8.x |
19:37.22 | [TK]D-Fender | vooduhal : Can you please clarify your term "MIB"? |
19:37.28 | Qwell[] | so far behind |
19:37.31 | vooduhal | SNMP mib database. |
19:37.32 | Damin | DAMN!!!! And you say that is going to be about $140 retail? |
19:37.38 | [TK]D-Fender | Qwell : With Cisco, 20th time's the charm ;) |
19:37.41 | Qwell[] | :p |
19:37.51 | [TK]D-Fender | Damin : Wicked cool ain't it? :D |
19:37.54 | Qwell[] | add a factor of 10, and you'd be pretty close |
19:38.01 | Damin | [tk] Yep.. |
19:38.22 | [TK]D-Fender | Qwell : 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-Fender | Damin : Category killer stuff.... |
19:38.56 | [TK]D-Fender | Damin : No normal user could need more than that phone. |
19:39.05 | [TK]D-Fender | Damin : And at Polycom quality. |
19:39.09 | Damin | stkn: Yep.. |
19:39.24 | Damin | [TK]D-Fender: Yep.. it's my new favorite phone. ;) |
19:39.38 | Damin | Anyone 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-Fender | Damin : Rest assured I will get one immediately upon its release. |
19:39.47 | Qwell[] | but no minibrowser...bah |
19:39.50 | Qwell[] | I'm not thrilled |
19:40.02 | [TK]D-Fender | Damin : Works great, I've got 2 fully loaded for my receptionist |
19:40.05 | vooduhal | So no ideas? |
19:40.09 | Damin | I've got a live-answer office application and that thing just looks awesome! :) |
19:40.13 | [TK]D-Fender | Qwell : And how many people seriously care about that? |
19:40.24 | CunningPike | Damin: afaik, the Attendant Console is just a 601 and a bunch of sidecars........ |
19:40.52 | Qwell[] | 1 at least |
19:41.02 | Qwell[] | coming from cisco, it's obligitory :p |
19:41.12 | SplasPood | Heh, anyone try the new cisco 7960 firmware 8.3 yet? |
19:41.15 | [TK]D-Fender | CunningPike : Actually the term applies to the sidecar alone. |
19:41.45 | CunningPike | [TK]D-Fender: Didn't know they worked by themselves....... |
19:41.48 | SplasPood | Don't |
19:41.52 | SplasPood | if you haven;t |
19:41.56 | SplasPood | it seems to have.. issues |
19:42.05 | Qwell[] | SplasPood: so don't use sip |
19:42.09 | C4T3l | does anyone know what the newest sip.ld version is on the polycom 601? |
19:42.09 | SplasPood | Damin: Yes, it does |
19:42.14 | Qwell[] | I'll be the skinny firmware works great. ;) |
19:42.17 | SplasPood | Damin: the polycom.. as of firmware 1.6.6 |
19:42.20 | Qwell[] | bet* |
19:42.23 | [TK]D-Fender | CunningPike : 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) |
19:42.31 | SplasPood | Damin: it always worked before, but was limited to 7 monitored extens |
19:42.41 | SplasPood | C4T3l: 1.6.6 |
19:43.10 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
19:43.21 | C4T3l | SplasPood: thanks. I'm trying to get presence working with the extension module |
19:43.26 | [TK]D-Fender | SplasPood : 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-Fender | C4T3l : it works very well here. |
19:43.47 | *** join/#asterisk TripleFFFFFFFFFF (n=TripleFF@147-102.mc.cite.net) |
19:43.49 | SplasPood | C4T3l: 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 |
19:43.55 | TripleFFFFFFFFFF | hey |
19:43.58 | Hmmhesays | i love that episode of voyager |
19:44.08 | SplasPood | [TK]D-Fender: See above.. I already said that to Damin |
19:44.09 | Damin | Cool.. so we'll be able to set status on phones and the attendant will visually see if a person is available or not? |
19:44.13 | TripleFFFFFFFFFF | is there a way to force a codec in the dialplan for a certain NPA ? |
19:44.16 | SplasPood | Damin: yup! |
19:44.19 | [TK]D-Fender | I had them since december without proper functionality. * adding SIP-B support for 1.4 will make them that much more powerful. |
19:44.22 | SplasPood | up to 48 hints now |
19:44.23 | C4T3l | SplasPood: thanks. how many with the 601? |
19:44.34 | Assid | 1.6.6 isnt public yet |
19:44.37 | SplasPood | C4T3l: 48 |
19:44.41 | SplasPood | Assid: yes it is |
19:44.46 | Assid | it is ?!?!? |
19:44.52 | SplasPood | Assid: its just not OLD so its not on the site /wo access |
19:44.57 | C4T3l | SplasPood: oh, with multiple mods |
19:45.09 | SplasPood | C4T3l: yes, or via the on screen buddy list |
19:45.32 | TripleFFFFFFFFFF | ?? |
19:45.33 | Assid | i have access.. but i dont think they gave me enough access to download sip updates |
19:45.38 | [TK]D-Fender | SplasPood : Yeah, but the buddy list browser is a totally ass way of doing it... |
19:45.51 | SplasPood | [TK]D-Fender: true, but it is possible :) |
19:46.03 | C4T3l | where could i get 1.6.6 |
19:46.08 | Qwell[] | C4T3l: Your reseller |
19:46.13 | SplasPood | C4T3l: Who'd you bu.. what Qwell said |
19:46.15 | [TK]D-Fender | SplasPood : A handset rectal exam is possible but I wouldn't suggest it ;) |
19:46.52 | TripleFFFFFFFFFF | so i guess its not possible ? |
19:46.53 | C4T3l | SplasPood: not sure. my boss just kinda threw it at me and said make it work |
19:47.46 | Assid | SplasPood: i go to voice downloads.. |
19:47.49 | Assid | but no file :( |
19:48.04 | TripleFFFFFFFFFF | you r sure no file ? |
19:48.09 | TripleFFFFFFFFFF | ;) |
19:48.14 | Assid | nah |
19:48.25 | TripleFFFFFFFFFF | hehe s there a way to force a codec in the dialplan for a certain NPA ? someone ? |
19:49.04 | Assid | can i grab 1.6.6 of someone here? |
19:49.16 | C4T3l | me too |
19:49.27 | C4T3l | or is that a violation of some law? |
19:49.41 | *** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk) |
19:50.31 | SplasPood | Yea its totally not legit, I believe |
19:50.43 | SplasPood | Although on the other hand I don't see why it SHOULD be.. |
19:51.40 | sb_mx | evening 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 |
19:51.59 | sb_mx | we've been toying around with chan_sip in order to prevent this |
19:52.24 | sb_mx | but we're not sure it a) is stable or b) makes sense |
19:52.27 | SplasPood | sb_mx: whats wrong with reloading all the extensions? |
19:53.11 | sb_mx | SplasPood, when you have more than 100 extensions with different contexts, it takes a lot of time to reload all the extensions/globals |
19:53.25 | SplasPood | I suppose |
19:54.04 | sb_mx | so we were actually thinkin of doing something like "sip reload 123" instead of just doing "sip reload" from the cli |
19:54.29 | SplasPood | how would you deal with a removal? |
19:54.34 | flujan | someone already configure a mfc/r2 E1? |
19:54.57 | sb_mx | SplasPood, in that case you'll have to do sip reload |
19:55.04 | TripleFFFFFFFFFF | as well as when you have 50,000 sip entries.. sip reload is nasty |
19:55.16 | SplasPood | realtime? |
19:55.54 | SplasPood | TripleFFFFFFFFFF: and whyy would one scale to 50k sip.conf entries on 1 box? |
19:56.16 | sb_mx | well, 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) |
19:57.17 | TripleFFFFFFFFFF | realtime |
19:57.54 | TripleFFFFFFFFFF | 100 clusterd boxes could have 50,000 entries shared |
19:57.58 | TripleFFFFFFFFFF | or not so shared |
19:58.33 | TripleFFFFFFFFFF | i think theres a big overhead in the qualify statement etc.. when over 300 entries |
19:58.51 | TripleFFFFFFFFFF | lots of crap going on.. but that my 0.00009 per min toughts |
20:01.47 | sb_mx | anyways, 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) |
20:06.41 | achandra | hello.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) |
20:09.16 | MattH | Hi... is thereanyway I can 'spoof' a BLF to a phone? |
20:09.23 | MattH | that being can I send a BLF notification (on or off) from the dialplan? |
20:10.12 | Hmmhesays | anyone know how to do a port range in iptables |
20:10.23 | Hmmhesays | i don't remember |
20:10.25 | docelmo | carefully |
20:10.31 | *** join/#asterisk ikey (n=er@203.115.29.2) |
20:10.34 | Hmmhesays | yeah no doubt |
20:10.46 | Hmmhesays | now for a useful answer |
20:10.49 | docelmo | I did it once and ended up having to reinstall linux |
20:10.49 | Hmmhesays | ... |
20:10.50 | Qwell[] | : |
20:10.56 | Qwell[] | I thought |
20:11.05 | Hmmhesays | i meant forward port ranges |
20:11.11 | Qwell[] | 10:20 |
20:12.39 | *** join/#asterisk TUplink (n=Tommy@68-232-82-147.chvlva.adelphia.net) |
20:12.52 | TUplink | can you sent instant messages over SIP with asterisk |
20:14.15 | file | TUplink: hello to you too |
20:14.30 | C4T3l | iptables -I FORWARD -d <ip> --dport 10:1000 -j KILLBOSS |
20:14.49 | achandra | lol |
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) |
20:15.16 | achandra | the 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) |
20:15.35 | C4T3l | it 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) |
20:24.57 | slobberknocker | is 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. |
20:26.12 | dlynes_office | slobberknocker: yeah...reload |
20:26.19 | dlynes_office | slobberknocker: or reload app_voicemail.so |
20:26.38 | slobberknocker | and that will not affect active channels or anything? |
20:26.53 | dlynes_office | slobberknocker: reload app_voicemail.so though does a complete reload; it doesn't just reload the config file |
20:27.03 | sb_mx | slobberknocker, also, you should try typing 'help' from the cli that'll give you some commands and their explanation |
20:27.35 | dlynes_office | slobberknocker: reload app_voicemail.so will probably affect current channels that are using voicemail. yes |
20:27.37 | slobberknocker | yeah, 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) |
20:28.01 | dlynes_office | slobberknocker: what does the conference call have to do with voicemail? |
20:28.09 | slobberknocker | nothing... |
20:28.18 | *** part/#asterisk terrapen (n=cjs@166.70.183.108) |
20:28.19 | slobberknocker | i am making changes to voicemail and need to reload the config |
20:28.22 | slobberknocker | but i have users on calls |
20:28.28 | dlynes_office | slobberknocker: Just use reload |
20:28.43 | dlynes_office | slobberknocker: Then you don't have to worry about whether anyone's currently logged into voicemail or not |
20:28.47 | achandra | that wont kick users? |
20:28.51 | dlynes_office | achandra: no |
20:28.52 | slobberknocker | sweet!!! thanks |
20:28.54 | sb_mx | slobberknocker, doing a reload or relaod app_voicemail.so shouldnt wont current channels. the channels will be "affected" from the next call on |
20:29.09 | slobberknocker | ok |
20:29.10 | sb_mx | slobberknocker, wont affect |
20:29.15 | achandra | good to know.. |
20:29.17 | slobberknocker | thanks |
20:29.26 | achandra | ;) good question |
20:29.39 | slobberknocker | its about time i dont as a stupid one |
20:29.45 | dlynes_office | sb_mx: reload app_voicemail.so will affect current channels that are accessing voicemail though, won't it? |
20:29.51 | slobberknocker | stupid 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) |
20:31.48 | sb_mx | dlynes_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) |
20:32.33 | dlynes_office | sb_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) |
20:33.31 | sb_mx | dlynes_office, i think so, yes. although im not one of the coders so whatever i tell you is from hand-on experience only |
20:34.06 | dlynes_office | sb_mx: yeah...well, that's how i gained knowledge of the reload command...watching it, and realizing that it didn't boot peeps :) |
20:34.08 | supjigatr | Any seen any virtual PBX dialplan examples, docs or ideas? |
20:34.32 | dlynes_office | sb_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 |
20:34.44 | docelmo | OI! |
20:34.49 | docelmo | err somethin |
20:35.02 | sb_mx | dlynes_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) |
20:35.23 | dlynes_office | supjigatr: what exactly are you trying to do? |
20:36.08 | supjigatr | Well 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. |
20:36.53 | supjigatr | I'm looking for a good model to follow or just bounce some ideas around. |
20:37.24 | dlynes_office | ah |
20:38.01 | Delta239 | any of you running a predictive or an autodialer? |
20:38.12 | s0lid | supjigatr, 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) |
20:38.40 | dlynes_office | Delta239: you're looking for documentation on it? |
20:38.58 | Delta239 | yeah.. if you have some please |
20:39.09 | dlynes_office | Delta239: do a search on voip-info.org for 'call file' |
20:39.46 | *** join/#asterisk somegeek (i=levin@unaffiliated/somegeek) |
20:39.49 | supjigatr | s0lid: well handling multi MOH, call limits etc. |
20:40.16 | Delta239 | lets see |
20:40.54 | sulan | How 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? |
20:41.28 | supjigatr | S0lid: Are you doing virtual PBX? |
20:42.36 | s0lid | not exactly PBX |
20:42.43 | s0lid | we are running inbound and outbound call centers |
20:42.43 | *** part/#asterisk slobberknocker (n=ckwall@63.149.122.94) |
20:43.02 | supjigatr | Ah. |
20:43.18 | supjigatr | We have a large call center 480 phones. |
20:43.37 | supjigatr | But we also have been dabbling in the small 4-6 phone virtual PBX. |
20:43.53 | supjigatr | I'm working on flowcharting out a good dialplan design. |
20:46.35 | Netgeeks | . |
20:46.42 | asterboy | I need help hooking up this phone to my * box: |
20:46.45 | asterboy | http://cgi.ebay.com/Vintage-Fisher-Price-chatter-telephone-C-1961_W0QQitemZ6059862027QQcategoryZ374QQrdZ1QQcmdZViewItem |
20:47.25 | asterboy | All it does is move the eyes when I dial. :( |
20:47.50 | *** join/#asterisk tdonahue-laptop (n=tdonahue@64.201.13.51) |
20:48.11 | dlynes_office | asterboy: that's because it's not a real phone |
20:48.18 | asterboy | dam |
20:48.30 | dlynes_office | asterboy: i used to have one of those when i was a kid :0 |
20:48.35 | asterboy | lol |
20:48.49 | asterboy | So much for selling those in the office. |
20:49.11 | dlynes_office | asterboy: well, if you bought it, you would know it's not a real phone, you dork |
20:49.31 | dlynes_office | asterboy: there's no phone cord on it |
20:49.52 | asterboy | ya I know, just kiddin around. |
20:49.56 | supjigatr | How do I convert this to the new CUT sytax? exten => _9NXXXXXX,1,Cut(channeltype=CHANNEL,/,1) |
20:49.58 | asterboy | Hey how about this one then! |
20:50.01 | asterboy | http://cgi.ebay.com/Vintage-Rainbow-Trout-Telephone-for-Den-w-Reels-Rods_W0QQitemZ7243545673QQcategoryZ793QQrdZ1QQcmdZViewItem |
20:50.55 | dlynes_office | what an uuuuuuuuuugly phone |
20:51.15 | asterboy | great gift for ya special gal |
20:51.24 | dlynes_office | oh yeah definitely |
20:51.43 | dlynes_office | i think she'd much rather have that than a nice meal at a fancy restaurant :p |
20:52.58 | Qwell[] | 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) |
20:54.48 | docelmo | Vonage goes public today and they are down over $2 from IPO |
20:54.51 | *** join/#asterisk viLeR (i=1000@200.114.70.228) |
20:54.56 | kuku5 | For some reason sip isnt going through my firwall ( cisco ), any suggestions on how to filter it to see whast happening ? ( outgoing calls work fine ) |
20:54.57 | docelmo | Their stock is gonna tank |
20:55.00 | dlynes_office | hahhaaha |
20:55.17 | SpaceBass | thats kind of a bad thing for VoIP in general |
20:55.25 | sb_mx | kuku5, maybe dnat to your * box? |
20:55.59 | kuku5 | dnat? |
20:56.10 | kuku5 | <PROTECTED> |
20:56.18 | kuku5 | for 5060 - 5063 udp |
20:56.43 | sb_mx | kuku5, are you somehow returning the traffic from the firewall to your * box? |
20:57.17 | asterboy | Wonder if these support SIP? |
20:57.19 | asterboy | http://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) |
20:58.11 | SpaceBass | asterboy, yeah but you have provide the 1s and 0s |
20:58.27 | asterboy | lol...talk in binary |
20:59.00 | SpaceBass | you know what they say....there are only 10 kinds of people, those who understand binary and those who dont |
20:59.05 | kuku5 | sb_mx: Im forwarding from the firewall to * |
20:59.08 | boch | what is the acceptable latency between an asterisk and a sipura to establish 2 telephone calls ? |
20:59.23 | dlynes_office | docelmo: dood....that totally rocks |
20:59.35 | dlynes_office | docelmo: pretty soon we won't have to compare our rates to vonage |
21:00.11 | docelmo | Who? |
21:00.16 | sb_mx | kuku5, are you missing sound or traffic to from incoming calls |
21:00.18 | docelmo | Who's rates |
21:00.22 | docelmo | I never did.. |
21:00.25 | asterboy | ya, I like that joke...but what about the other 9 people? :P |
21:00.27 | docelmo | I was always cutting edge |
21:00.28 | dlynes_office | docelmo: anyone's rates |
21:00.34 | kuku5 | traffic - the call doesnt get to it, broadvoice things its busy |
21:00.38 | dlynes_office | docelmo: you mean same rate as vonage? |
21:00.52 | docelmo | Nope.. Im MUCH cheaper across the board |
21:00.52 | sulan | I 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.. :( |
21:00.54 | Damin | boch: 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. |
21:01.06 | dlynes_office | docelmo: even for unlimited minutes? |
21:01.48 | docelmo | I would have have to check our retail platform |
21:02.13 | dlynes_office | docelmo: yeah...we can beat vonage at everything except the unlimited packages |
21:02.21 | dlynes_office | docelmo: those packages, we can't even come close |
21:02.42 | docelmo | Im not sure what were selling for honestly.. Im more BYOD/Wholesale |
21:03.12 | dlynes_office | docelmo: $30/mo unlimited north america |
21:03.25 | dlynes_office | docelmo: $40/mo unlimited north america for businesses |
21:03.42 | dlynes_office | docelmo: and that's $Cdn |
21:03.44 | docelmo | hmmm brb Im gonna go have a look see |
21:03.48 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219) |
21:04.16 | docelmo | $25USD unlimited US/Canada |
21:04.39 | docelmo | Wanna buy some A-Z wholesale? |
21:04.42 | docelmo | :) |
21:04.47 | sevard | how much to liberia |
21:04.58 | _alex_mx | isn't US/Canada in north america? |
21:05.05 | *** join/#asterisk AuPix (n=AuPix@adsl-04-85.abel.net.uk) |
21:05.10 | dlynes_office | _alex_mx: yes |
21:05.18 | docelmo | no |
21:05.24 | _alex_mx | so is it 25 or 30 |
21:05.28 | dlynes_office | _alex_mx: as is mexico |
21:05.29 | sevard | dlynes_office: how much for Liberia |
21:05.38 | docelmo | Mexico is central |
21:05.53 | dlynes_office | docelmo: ah...maybe in American geography |
21:06.00 | sevard | Mexico is still considered north in most cases |
21:06.11 | _alex_mx | uhmmm not, which is why it's called the north american free trade agreement |
21:06.11 | dlynes_office | docelmo: in Canadian geography, we count it as North America |
21:06.15 | docelmo | whatever.. I hate geography |
21:06.20 | sevard | heh |
21:06.21 | sevard | so |
21:06.22 | sevard | um |
21:06.23 | sevard | Liberia? |
21:06.34 | docelmo | Where the hell is liberia? |
21:06.37 | dlynes_office | sevard: we don't do wholesale |
21:06.39 | sevard | FUCKING AFRICA |
21:06.40 | sevard | jk |
21:06.42 | docelmo | ohh |
21:06.43 | sevard | but seriously |
21:06.45 | sevard | africa. |
21:06.45 | dlynes_office | North Africa |
21:06.58 | *** join/#asterisk aze_ (n=aze@ACayenne-101-1-6-117.w81-248.abo.wanadoo.fr) |
21:07.01 | dlynes_office | Borders the Mediterranean |
21:07.02 | sevard | docelmo: did you say you do A-Z? |
21:07.10 | docelmo | check out http://www.plainvoip.com/?page=showrates |
21:07.15 | sulan | am 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? |
21:07.18 | sevard | i like my voip plain. |
21:07.22 | sevard | vanilla |
21:07.22 | docelmo | BYOD and wholesale |
21:07.41 | sevard | non-BYOD services are uber-gay |
21:07.45 | sevard | i would NEVER buy |
21:07.54 | Hmmhesays | shut up |
21:07.55 | docelmo | You probably dont have the minutes to qualify |
21:08.04 | SpaceBass | hey Hmmhesays whats happenin' |
21:08.15 | Hmmhesays | working with idiots |
21:08.16 | Hmmhesays | you? |
21:08.30 | SpaceBass | just finished working with idiots |
21:08.35 | Hmmhesays | I got really drunk last night and was 1 1/2 late for work |
21:08.35 | dlynes_office | sulan: you're accepting connections from any sip client? |
21:08.38 | SpaceBass | not im working up the mental will power to go buy beer |
21:08.39 | docelmo | Im leaving my idiots now.. |
21:08.43 | docelmo | cya @ home.. |
21:09.03 | sulan | dlynes_office: well, I thought it would be good if you could call the office without going through PSTN first.. :) |
21:09.04 | sevard | i'm having major pipe issues visiting plainvoip.com |
21:09.20 | dlynes_office | sulan: use a username and password to access it |
21:09.29 | dlynes_office | sulan: otherwise you're asking for trouble |
21:09.56 | dlynes_office | sulan: also force them to do sip registrations |
21:10.23 | sulan | dlynes_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 |
21:10.27 | dlynes_office | sulan: if you don't force them to do sip registrations, every user can borrow someone else's username and password |
21:11.06 | dlynes_office | and a 407 is what? |
21:11.18 | sulan | 407 Proxy Authentication Required |
21:11.35 | sulan | in essence, Asterisk tries to authenticate the remote user because of the username clash |
21:11.43 | dlynes_office | sulan: and that's a bad thing? |
21:12.21 | sulan | dlynes_office: well, then SIP clients with the local username foo can't call the office, even though they're registered at a foreign registrar |
21:12.33 | SpaceBass | yep, off to buy beer |
21:12.33 | SpaceBass | peace |
21:12.41 | Hmmhesays | have one for me SpaceBass |
21:13.03 | dlynes_office | sulan: 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) |
21:13.53 | sulan | dlynes_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... |
21:15.18 | sulan | "When calling us, make sure your SIP-provider hasn't given you any of these usernames: Alice, Bob, Foo, Bar"... |
21:15.41 | dlynes_office | sulan: 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? |
21:16.04 | sulan | dlynes_office: no... that's not what I'm trying to say... :) |
21:16.40 | dlynes_office | sulan: can you reword it so you're a little more clear, then? |
21:17.27 | sulan | Lets 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) |
21:17.48 | dlynes_office | ok |
21:18.21 | dlynes_office | go on |
21:18.27 | sulan | then 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 |
21:19.05 | sulan | when 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) |
21:19.58 | dlynes_office | and so that user that can't authenticate themselves is not able to access your autoattendant, right? |
21:20.03 | sulan | only 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 |
21:21.01 | sulan | dlynes_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 |
21:22.13 | dlynes_office | try posting something to bugs.digium.com? |
21:22.23 | dlynes_office | check to make sure your issue isn't already there |
21:22.29 | sulan | sure... |
21:23.35 | sulan | Matching incoming calls to users and peers |
21:23.36 | sulan | Asterisk 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. |
21:23.42 | sulan | http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+type |
21:24.57 | sulan | You would think that it's intended when you read that paragraph... but it must be very bad, considering the scenario I have described? |
21:29.05 | kuku5 | Anyone willing to debug sip with me? |
21:29.26 | kuku5 | im having some nat isssues, outgoing calls dont want to come in |
21:29.26 | sevard | hahah |
21:29.27 | sevard | yeah right |
21:29.43 | kuku5 | :) |
21:31.12 | *** join/#asterisk qdk (n=qdk@x1-6-00-0f-66-90-6b-48.k441.webspeed.dk) |
21:33.35 | [TK]D-Fender | kuku5 : Pastebin your sip.conf |
21:34.13 | vader-- | hello |
21:34.23 | vader-- | hows it going guys |
21:38.21 | SpaceBass | sometimes apple is stupid...i just needed to get that out |
21:38.29 | [TK]D-Fender | Qwell, 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-Fender | SpaceBass : More like a LOT. |
21:38.59 | *** join/#asterisk hypnox (n=dan@cornelyn.force9.co.uk) |
21:39.09 | SpaceBass | Im usually a big fan/supporter but I hate that you cannot buy songs from other counteries' music stores |
21:39.24 | hypnox | can anyone think why all the messages in the console might be appearing twice? It only happens after I do a reload. |
21:39.30 | dlynes_office | SpaceBass: fruit doesn't have brains...how could an apple possibly be smart? |
21:39.41 | SpaceBass | should have seen that coming |
21:40.30 | dlynes_office | hypnox: you've got two loggers defined in your logger.conf file? |
21:40.49 | dlynes_office | hypnox: for console messages, that is? |
21:41.16 | dlynes_office | SpaceBass: btw...that's a thing that certain products have to protect their markets |
21:41.39 | SpaceBass | i guess...but isnt a sale a sale? |
21:42.06 | hypnox | dlynes_office hmm as far as i can tell my logger.conf is still as the default one |
21:42.13 | dlynes_office | SpaceBass: 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 |
21:42.31 | SpaceBass | so can I not buy a car in canada? |
21:42.47 | dlynes_office | SpaceBass: sure, you can...but you have to pay all the duties to get it back across |
21:42.53 | SpaceBass | gotcha |
21:43.07 | dlynes_office | SpaceBass: 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 |
21:43.21 | Delta239 | need some hel |
21:43.22 | dlynes_office | SpaceBass: or at least that used to be the case |
21:43.25 | Delta239 | help |
21:43.28 | Delta239 | on this website |
21:43.30 | Delta239 | http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out |
21:43.38 | [TK]D-Fender | And conversly Canadian weather wrecks cars so much fast than US cars and they last less.... |
21:43.39 | Delta239 | i am trying to do the example 2 |
21:43.43 | SpaceBass | well all I wanted was to buy some stinkin' french music! |
21:44.27 | dlynes_office | SpaceBass: from a website in france? |
21:44.32 | docelm0 | stinking is about right |
21:44.35 | SpaceBass | itunes music store |
21:44.42 | dlynes_office | SpaceBass: in france? |
21:45.03 | dlynes_office | SpaceBass: key word here being 'france'? |
21:45.04 | Delta239 | and is not working |
21:45.21 | SpaceBass | won't let you if your credit card billing address is outside of france |
21:45.28 | SpaceBass | and I dont have a bank account there anymore |
21:45.29 | SpaceBass | oh well |
21:45.34 | hypnox | hmm this is wierd |
21:45.39 | dlynes_office | SpaceBass: Yeah, anyways...it could be that way for two reasons |
21:45.53 | dlynes_office | SpaceBass: either the people that designed the website didn't have enough forethought |
21:46.19 | dlynes_office | SpaceBass: 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) |
21:46.35 | SpaceBass | actually, its apple...and i'm sure it has to do with royalties, etc |
21:46.44 | SpaceBass | and the record companies |
21:47.01 | dlynes_office | SpaceBass: can't you buy the french music anywhere else, then? |
21:47.24 | SpaceBass | i can find it elsewhere....i was just venting |
21:47.29 | *** part/#asterisk _alex_mx (n=_alex_mx@200.94.154.226) |
21:47.37 | dlynes_office | SpaceBass: like maybe itunes.ca or itunes.ag, or any other french country? |
21:47.48 | Delta239 | brb |
21:48.04 | SpaceBass | apples 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) |
21:48.31 | Nugget | Alizeé shakes it. |
21:48.46 | dlynes_office | SpaceBass: ah...cute |
21:49.02 | SpaceBass | alors, Alizee est choud! |
21:49.18 | Nugget | heh |
21:49.25 | SpaceBass | didnt think she was still around actually |
21:49.27 | Nugget | easy on the eyes, hard on the ears. |
21:49.32 | Renacor | is 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? |
21:49.47 | SpaceBass | anyway...Im just trying to distract myself from packing up this WIP330 and mailing it back |
21:50.16 | dlynes_office | SpaceBass: have you tried searching on www.musiqueplus.com? |
21:50.33 | dlynes_office | SpaceBass: or www.muchmusic.com/tv/frenchkiss/? |
21:50.35 | *** join/#asterisk AlexCTI (n=alex@adsl-074-238-025-003.sip.mia.bellsouth.net) |
21:50.36 | SpaceBass | no, but I will, thanks |
21:50.41 | vader-- | any of oyu guys doing cdr to mysql? |
21:51.13 | dlynes_office | SpaceBass: 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) |
21:51.40 | dlynes_office | SpaceBass: both web sites are in canada, so there shouldn't be any cross-border problems if you're in the US |
21:51.42 | SpaceBass | pas problem |
21:51.55 | Nugget | nah, who needs to be able to read french? Just light up a gauloises and look disaffected. |
21:51.59 | dlynes_office | ne probleme pas? |
21:52.05 | SpaceBass | LOL |
21:52.38 | dlynes_office | Well, all the French here in Canada drink Labatt's Blue or Labatt's Blue Light |
21:52.57 | dlynes_office | It's because they can still ask for it when they're totally inebriated :) |
21:53.05 | dlynes_office | boo and boo li |
21:53.06 | *** join/#asterisk rg1_ (n=rg1@www.airlinksystems.com) |
21:53.36 | rg1_ | anyone know how I can get "monitor" to play a beep for the caller to know they are recording? |
21:53.55 | [TK]D-Fender | vous etes tous le pire des francophones dans canal sti! |
21:54.21 | Nugget | gros glandeur! |
21:54.25 | dlynes_office | tabernac! |
21:54.33 | [TK]D-Fender | sus mon pipe toi! |
21:54.49 | [TK]D-Fender | :D |
21:55.05 | dlynes_office | under your pipe? |
21:55.17 | dlynes_office | erm no wait...that'd be sous, not sus :) |
21:55.27 | SpaceBass | sous |
21:55.28 | SpaceBass | yeah |
21:55.38 | dlynes_office | i don't know what the hell sus is :0 |
21:56.01 | rg1_ | anyone know how I can get "monitor" to play a beep for the caller to know they are recording? |
21:56.23 | SpaceBass | rg1_, does it not make the 'beep' sound now? thought that was part of monitor |
21:56.41 | rg1_ | want it to beep like every 15 seconds or so |
21:56.50 | SpaceBass | ahhh |
22:00.47 | *** join/#asterisk scoody650 (n=name@h-68-165-169-170.snvacaid.covad.net) |
22:01.12 | scoody650 | hello |
22:01.15 | [TK]D-Fender | dlynes_office : So I type like shit, SHUP HOE! :D |
22:02.10 | [TK]D-Fender | dlynes_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) |
22:02.31 | scoody650 | is anyone familiar with setting up DIDs |
22:03.07 | [TK]D-Fender | scoody650 : Yes. |
22:03.56 | file | uh... oh... |
22:04.07 | scoody650 | great. 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-Fender | scoody650 : NO. |
22:04.30 | scoody650 | what card would i need |
22:04.34 | [TK]D-Fender | scoody650 : Analog lines do not support the concept of DID's |
22:04.41 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
22:04.41 | scoody650 | i didn't think so |
22:04.51 | [TK]D-Fender | scoody650 : And Digital trunk. |
22:04.54 | dlynes_office | [TK]D-Fender: no idea what you said...but then again, i'm not french :0 |
22:05.05 | [TK]D-Fender | dlynes_office : Which part? |
22:05.28 | dlynes_office | [TK]D-Fender: the whole diminished major shit...that sounded more like music theory than language theory |
22:05.32 | [TK]D-Fender | dlynes_office : And FYI I just speak the language, I'm english raised. |
22:05.41 | [TK]D-Fender | dlynes_office : I *was* talking music :) |
22:05.59 | dlynes_office | [TK]D-Fender: ah...you were talking chords :) |
22:05.59 | scoody650 | [TK]D-Fender: Which card would you recommend for using DID's |
22:06.06 | dlynes_office | choard! :) |
22:07.02 | [TK]D-Fender | dlynes_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-Fender | scoody650 : Any digital card. And of Digium's TE line, or Sangoma's A1/2/4 lines |
22:08.09 | scoody650 | so 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) |
22:09.06 | ManxPower | does anyone have a sound file of Allison saying "If you are sending a fax, press "start" now." |
22:09.06 | SpaceBass | a DID is a term that describes a number that can be dialed from outside your PBX that connects to your PBX |
22:09.08 | SpaceBass | more or less |
22:09.10 | SpaceBass | in a round about way |
22:09.24 | scoody650 | right |
22:09.35 | ManxPower | DID is the same as DDI and means "telephone number direct to your extension, not requring an operator or an IVR" |
22:09.39 | ManxPower | ~did |
22:09.41 | jbot | did is, like, Direct Inward Dialing |
22:09.47 | ManxPower | ~ddi |
22:09.48 | jbot | from memory, ddi is Direct Dialling Inward, URL: http://www.wilco-telephony.co.uk/did.html |
22:10.11 | scoody650 | does it come from the telco as a RJ45 interface? |
22:10.20 | scoody650 | it looks like the TE cards are RJ45 |
22:10.21 | [TK]D-Fender | scoody650 : 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. |
22:10.42 | ManxPower | There are at least 5 ways to do DID/DDI. |
22:10.49 | ManxPower | The most common is using a PRI |
22:11.03 | dlynes_office | Does anyone know a way to get outbound calls to show up on zap show channels? |
22:11.16 | ManxPower | dlynes_office, they already do |
22:11.26 | dlynes_office | ManxPower: on a pri? |
22:11.27 | [TK]D-Fender | scoody650 : Before you start getting in way over your ahead and to far ahead of yourself, what do you use NOW? |
22:11.28 | ManxPower | Ahrimanes, no they don't, they show up in "show channels" |
22:11.47 | dlynes_office | ManxPower: yeah, exactly |
22:11.57 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-51-95.cybersurf.com) |
22:12.01 | dlynes_office | ManxPower: 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) |
22:12.03 | scoody650 | [TK]D-Fender: This is a new system. AT&T reccomended a DID setup for my incoming calls |
22:12.20 | dlynes_office | ManxPower: incoming calls show up on zap show channels, but outbound don't |
22:12.25 | scoody650 | [TK]D-Fender: and two outbound POTS lines |
22:12.26 | Renacor | Is there a command that you can have asterisk do a zapbarge into a channel, then call an extension to let you listen in? |
22:12.32 | ManxPower | scoody650, I'm happy for you. Now what type of line does AT&T recommend? |
22:12.44 | ManxPower | Renacor, no, but you could write one |
22:13.03 | scoody650 | ManxPower: I'm not sure i understand the question |
22:13.13 | ManxPower | Recommending "a DID line" is about as useful as recommending "a Caller*ID line" |
22:13.49 | vader-- | 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) |
22:14.20 | ManxPower | scoody650, 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". |
22:14.41 | CrashHD | anyone have a website with standard us *codes |
22:14.41 | ManxPower | ..and 100 DIDs, consecutive" |
22:14.45 | scoody650 | thanks, i think that makes more sense |
22:14.47 | CrashHD | likes *69 etc... |
22:14.54 | scoody650 | they quoted me two DID trunks |
22:15.03 | ManxPower | CrashHD, you mean line the one on the Wiki, which was copied from NANPA |
22:15.06 | dlynes_office | CrashHD: they're standard codes, period |
22:15.10 | ManxPower | scoody650, you do not want those. |
22:15.16 | scoody650 | really, alright |
22:15.19 | dlynes_office | CrashHD: Just do a google search for NANPA vertical service codes |
22:15.27 | CrashHD | nanpa is the keyword I was looking for |
22:15.27 | CrashHD | thanks |
22:15.34 | ManxPower | sounds like they are quoting you inward only DID lines on either analog or Channelized T-1. |
22:15.51 | scoody650 | i think it'll be on analog since i'm not getting a T1 |
22:16.01 | CrashHD | scoody650: 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) |
22:16.02 | ManxPower | Ah. You can't use those lines with Asterisk then |
22:16.05 | *** join/#asterisk Qwell (n=north@unaffiliated/qwell) |
22:16.14 | CrashHD | did trunks can not be analog |
22:16.17 | scoody650 | ok, so what kind of lines then. the PRI? |
22:16.21 | ManxPower | digium analog cards do not support DID |
22:16.32 | ManxPower | CrashHD, DID trunks CAN be analog |
22:16.44 | ManxPower | Usually delivered over E&M Wink Analog |
22:16.51 | CrashHD | news to me |
22:17.07 | [TK]D-Fender | scoody650 : What do you have NOW? |
22:17.18 | dlynes_office | what's the diff between an analog trunk and a did analog trunk? |
22:17.22 | scoody650 | i have nothing but a punch block in a closet |
22:17.31 | scoody650 | this is a fresh installation |
22:17.35 | ManxPower | CrashHD, E&M Wink is an analog signaling method. They just adapted it for use on T-1 |
22:17.50 | [TK]D-Fender | scoody650 : Ok, so you are building a PBX for a company that doesn't HAVE one right now? |
22:18.03 | scoody650 | correct |
22:18.05 | ManxPower | dlynes_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-Fender | scoody650 : how many users? |
22:18.31 | dlynes_office | ManxPower: so a did analog trunk is like an analog trunk, but you can deliver multiple dids over it? |
22:18.32 | scoody650 | 4 to start, but no more than twenty for a while |
22:18.57 | dlynes_office | ManxPower: i.e. it's effectively like a pri on a single analog line? |
22:19.10 | ManxPower | dlynes_office, It's just a type of signaling. |
22:19.16 | hinckc | dlynes_office: always impossible to say without knowing who's saying it (each equipt mfg abuses terms differently), but the difference may be CallerID |
22:19.17 | ManxPower | noky, nothing like a PRI |
22:19.18 | CrashHD | it has to use out of band dtmf or whatever to pass the dids right? |
22:19.40 | ManxPower | CrashHD, there is no such thing as out of band DTMf on analog |
22:19.41 | [TK]D-Fender | ManxPower : 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) |
22:20.25 | ManxPower | usually it's sent as something like Telco: WINK. PBX: Wink. TELCO; DID DTMF or DID PULSE |
22:20.35 | scoody650 | [TK]D-Fender: i will only be starting with 4 users |
22:20.36 | dlynes_office | CrashHD: isn't out-of-band a purely digital concept? |
22:20.43 | ManxPower | I'll bet you didn't know that telcos can send the DID digits using PULSE DIALING |
22:20.44 | CrashHD | dlynes_office: ya |
22:21.04 | [TK]D-Fender | scoody650 : Ok, Forget the concept of DID, you're talking a handful of NORMAL lines at best for that install size. End of story. |
22:21.09 | CrashHD | dlynes_office: figured I'd difer to ManxPower. I don't deal with analog for much |
22:21.11 | ManxPower | dlynes_office, I would have to look at the actual specs to be more accurate. |
22:21.28 | [TK]D-Fender | scoody650 : or "plan B" of getting all of your lines provided by a VoIP provider which could offer DID's |
22:21.49 | hinckc | ManxPower: 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. |
22:21.54 | ManxPower | Some providers also do things like send DID-DTMF*CALLERID-DTMF# |
22:22.09 | scoody650 | [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-Fender | scoody650 : For 4 its nowhere near profitable for DID capabilities unless you go pure VoIP. |
22:22.38 | ManxPower | scoody650, If you want DID with Asterisk then you must use a T-1 |
22:22.50 | dlynes_office | scoody650: you could always get them a fractional pri later on |
22:22.59 | scoody650 | thanks, that nails that down |
22:23.01 | ManxPower | Or VoIP, but only a moron would send all their calls over VoIP |
22:23.02 | dlynes_office | scoody650: when they get big enough something like that's practical |
22:23.11 | scoody650 | so would i just get four phone numbers next to each other? |
22:23.21 | [TK]D-Fender | ManxPower : Not without a "plan B" failover... |
22:23.36 | ManxPower | [TK]D-Fender, that is where we disagree. |
22:23.46 | CrashHD | why wouldn't he just go channelized t1, that's cheap and easy and he could expand and get dids |
22:23.56 | CrashHD | thats what I would be doing |
22:24.04 | CrashHD | why screw with telco hunt groups |
22:24.05 | [TK]D-Fender | ManxPower : Truely awsome offering : manx |
22:24.11 | scoody650 | compared 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) |
22:24.13 | ManxPower | CrashHD, Channelized T-1 MAY or MAY NOT be cheap. |
22:24.15 | [TK]D-Fender | ManxPower : So not even with a failover hmm? |
22:24.23 | [TK]D-Fender | ManxPower : http://www.polycom.com/products_services/0,1443,pw-34-182-15672,00.html |
22:24.27 | Renacor | where is the zapbarge command written? |
22:24.30 | [TK]D-Fender | ManxPower : missed the link there. |
22:24.35 | ManxPower | [TK]D-Fender, I have my own source for Polycom |
22:24.41 | ManxPower | Hell, I'm Polycom certified. |
22:24.45 | ManxPower | for what that's worth. |
22:24.48 | dlynes_office | Renacor: /usr/local/src/asterisk-1.2.7.1/apps/app_zapbarge.c |
22:24.56 | [TK]D-Fender | ManxPower : About $1.50? ;) |
22:24.58 | Renacor | dlynes_office: thanks |
22:25.03 | ManxPower | does anyone have a sound file of Allison saying "If you are sending a fax, press "start" now." |
22:25.08 | dlynes_office | [TK]D-Fender: $1.50 for a polycom? |
22:25.24 | ManxPower | [TK]D-Fender, there's a certificate somwhere with my name on it. |
22:25.25 | [TK]D-Fender | dlynes_office : No, the worth of ManxPower's certification ;) |
22:25.27 | scoody650 | so should i purchase 4 analog lines and run them all into a TDM400P with FXo ports? |
22:25.42 | [TK]D-Fender | scoody650 : Do you even need 4 lines? |
22:25.53 | ManxPower | scoody650, If you were my customer I would tell you not to use Asterisk and use a dumb as rocks comdial or similar system |
22:25.58 | scoody650 | with four people i think so, to allow for inbound and outbound |
22:26.07 | dlynes_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 |
22:26.21 | dlynes_office | [TK]D-Fender: the customer doesn't know that |
22:26.40 | [TK]D-Fender | dlynes_office : I know.. and its great to be able to just grab resources yourself instead of having to ask like most other people :) |
22:26.43 | ManxPower | dlynes_office, I don't try to sell Polycoms to a customer. If a customer wants me to deploy VoIP then they WILL get Polycom |
22:26.44 | dlynes_office | [TK]D-Fender: same thing with MCP and MCSA and all that other crap |
22:27.18 | scoody650 | if i had four lines do i jsut have them hunt to the next line if the first is busy? |
22:27.25 | ManxPower | Actually 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-Fender | ManxPower : 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. |
22:28.03 | dlynes_office | ManxPower: and you can actually get them wholesale, then |
22:28.12 | ManxPower | I really should do their online training to get other polycom certs |
22:28.53 | [TK]D-Fender | scoody650 : yes, you would have them on a hunt group, and have their callerID all set to the primary # |
22:28.54 | ManxPower | I 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) |
22:29.05 | Renacor | damn, can't believe there isn't a spy command that will call an extension instead of you having to call |
22:29.11 | Nugget | don't you mean REALTOR®s? |
22:29.12 | *** join/#asterisk RF_MIA (n=unknown@adsl-070-147-214-250.sip.mia.bellsouth.net) |
22:29.16 | scoody650 | [TK]D-Fender: ok, that makes sense |
22:29.24 | ManxPower | Nugget, Something like that. |
22:29.30 | ManxPower | I have to deal with 350 of them |
22:29.33 | Nugget | oof |
22:29.36 | dlynes_office | Why is it nobody seems to know what unicode is? |
22:29.53 | Nugget | I made a shirt. |
22:29.59 | scoody650 | [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? |
22:30.35 | Nugget | dlynes_office: http://www.cafepress.com/nucleartacos.26746951 |
22:30.36 | dlynes_office | Nugget: Yeah, but your spanish or whatever language it is you're using, is showing up as garbage characters :) |
22:30.47 | Nugget | http://www.cafepress.com/nucleartacos.26721820 (for mac users) |
22:31.05 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
22:31.06 | ManxPower | dlynes_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? |
22:31.31 | dlynes_office | ManxPower: I see you're a good old boy from the sawth? |
22:31.36 | RF_MIA | unicode for R2? |
22:31.58 | ManxPower | Nugget, damn you! I thought you made a shirt about REALTORS(r) |
22:31.59 | [TK]D-Fender | scoody650 : sure |
22:32.08 | ManxPower | dlynes_office, Actually I'm a yankee |
22:32.15 | Nugget | sorry :) |
22:32.26 | scoody650 | great. that was my main fear. |
22:32.29 | dlynes_office | Nugget: so what language was it? |
22:32.34 | ManxPower | I 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 |
22:33.12 | dlynes_office | Nugget: 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 |
22:33.32 | dlynes_office | [hC]: first one is channel 1-24 |
22:33.43 | [hC] | Aha. |
22:33.43 | dlynes_office | [hC]: the second card is channel 25-48 |
22:33.44 | [hC] | Of course! |
22:33.46 | Nugget | Learn 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 |
22:34.02 | dlynes_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-Fender | every 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? |
22:34.47 | dlynes_office | [hC]: btw...i haven't even gotten around to setting any sangoma cards up yet :) |
22:35.05 | [hC] | hah |
22:35.08 | dlynes_office | [hC]: i can let you know by Friday |
22:35.11 | [hC] | i'm putting in my first a200's |
22:35.14 | dlynes_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 |
22:35.30 | dlynes_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-Fender | I'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. |
22:35.49 | RF_MIA | do 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? |
22:36.14 | dlynes_office | RF_MIA: We got an echo can with this one, so we don't look like dumb asses should the customer get echo |
22:36.32 | RF_MIA | good point dlynes |
22:36.33 | dlynes_office | RF_MIA: this way it's guaranteed they don't get echo |
22:36.38 | [TK]D-Fender | General question I could use a hand with : Which * package has the sample MP3's for MoH in it? |
22:36.45 | RF_MIA | "hopefully" they won't |
22:36.51 | scoody650 | thanks for the help everyone |
22:36.56 | RF_MIA | I doubt its guaranteed...echo is a strange beast |
22:37.00 | bkw__ | Does anyone have any more goodies on how to get a 7970G setup? the voip-info wiki sucks |
22:37.07 | dlynes_office | RF_MIA: yeah, but sangoma's is carrier grade |
22:37.15 | [TK]D-Fender | RF_MIA : DON'T. Sangoma's EC is pretty rock-solid |
22:37.27 | RF_MIA | good to know |
22:37.40 | file | it's the Ocstasic(sp?) chip |
22:37.41 | [TK]D-Fender | RF_MIA : You'd have to be in a SICK scenario to overload that thing... |
22:37.46 | [TK]D-Fender | file : yup |
22:37.47 | file | expensive, but works well |
22:38.09 | dlynes_office | file: yeah, but if you're popping it into a rural setting, what else are you going to use? |
22:38.12 | [TK]D-Fender | file : No more so than any other guys solution, a whole lot better integrated, and scales cheaper. |
22:38.19 | file | I meant the actual chip |
22:38.29 | file | not Sangoma product as a whole |
22:38.30 | supjigatr | The sangoma will not have any echo. |
22:38.34 | dlynes_office | file: it's either a digium card with a tellabs ec, or a sangoma with their ec |
22:38.59 | rg1_ | In asterisk, can someone help me with the syntax of how I can Goto a "label" instead of a "priority" ? |
22:39.01 | dlynes_office | file: both solutions are expensive |
22:39.02 | supjigatr | The tellabs card 64ms card with daughter works well too. |
22:39.28 | supjigatr | The a104d works great once you get it installed. |
22:39.30 | dlynes_office | but, with sangoma's solution, you buy the ec for your first card, and it handles all the remora daughterboards, too |
22:39.31 | supjigatr | 0 echo |
22:40.02 | [TK]D-Fender | rg1_ : Goto(label) |
22:40.21 | dlynes_office | so, it's only $230 or so for the ec, regardless of whether it's 2 lines, or 24 lines |
22:40.22 | rg1_ | and how would I specify the label |
22:40.26 | rg1_ | in the dialplan? |
22:40.33 | [TK]D-Fender | supjigatr : Yup... mine's been gold from the moment I installed it. |
22:40.48 | [TK]D-Fender | rg1_ : Go read THEBOOK. |
22:40.51 | [TK]D-Fender | ~thebook |
22:40.58 | dlynes_office | ~book |
22:41.00 | jbot | book is probably a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
22:41.02 | supjigatr | It 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. |
22:41.16 | rg1_ | thank you muchly |
22:41.27 | dlynes_office | supjigatr: doesn't work on slackware 10.1? |
22:41.41 | dlynes_office | supjigatr: or you mean with the kernel that comes with slackware 10.1? |
22:41.53 | [TK]D-Fender | supjigatr, dlynes_office : Yeah, I'd suspect kernel |
22:41.55 | dlynes_office | supjigatr: i.e. 2.4.37, instead of 2.4.39? |
22:42.01 | supjigatr | kernel and 10.1 udev setup |
22:42.19 | supjigatr | just upgrading kernel doesn't work. |
22:42.21 | dlynes_office | supjigatr: yeah...i've only ever used 2.4.39 or 2.6.15.5 |
22:42.44 | supjigatr | 10.2 just use test26.s and plug the card in. |
22:42.46 | dlynes_office | erm |
22:42.56 | dlynes_office | 2.4.27/2.4.29 |
22:43.00 | supjigatr | I have boxes of 411p if you wanna try digium. |
22:43.01 | dlynes_office | 2.4.26 was 10.0 |
22:43.22 | [TK]D-Fender | Slackware 11 is due out any time.... |
22:43.33 | [TK]D-Fender | wonder when Pat will finally cave to 2.6.... |
22:43.38 | dlynes_office | [TK]D-Fender yeah...stop teasing me |
22:43.51 | dlynes_office | [TK]D-Fender: it's been due out any time for about 4 months now |
22:44.00 | supjigatr | Hehe. Long as test26.s is there its easy to install with 2.6 |
22:44.24 | [TK]D-Fender | dlynes_office : Thats the Slackware way! |
22:44.34 | supjigatr | Hey on your 104d how are the ports labeld? |
22:44.39 | dlynes_office | [TK]D-Fender: all i know is when it's available |
22:44.50 | dlynes_office | [TK]D-Fender: i'll get it before any of you schleps :) |
22:44.56 | [TK]D-Fender | I still am scared to try and upgrade my kernel... I'm still a major linux newb... |
22:45.00 | dlynes_office | I've got a subscription :) |
22:45.33 | supjigatr | um make bzlilo. |
22:45.46 | dlynes_office | [TK]D-Fender: change the install_path in your make file |
22:45.48 | supjigatr | I just grab the config file from slack disk and use it if the bootdisk worked. |
22:46.15 | dlynes_office | [TK]D-Fender: then do make distclean ; make menuconfig ; make bzlilo ; make modules ; make modules_install |
22:46.32 | dlynes_office | [TK]D-Fender: erm do the install_path change after the make distclean |
22:46.38 | supjigatr | and if on slack make sure it puts vmlinuz where lilo thinks it is. |
22:46.50 | nahirean | anyone know why * would hang for incoming calls without RSA, but produce cause code 3 - no such context/extension for when RSA is enabled? |
22:48.42 | ManxPower | Agents are the biggest Drama Queens |
22:48.49 | [TK]D-Fender | dlynes_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) |
22:50.36 | dlynes_office | [TK]D-Fender: i do remote kernel upgrades on all my machines |
22:51.02 | dlynes_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-Fender | dlynes_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) |
22:55.52 | binhex | I 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? |
22:57.16 | dlynes_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) |
22:59.49 | achandra | binhex: what does the /etc/sysconfig/network for ifcfg-eth0 say in it ? |
23:00.18 | [TK]D-Fender | dlynes_office : Its my * server, file, FTP, web, HTPC, and used to make me coffee too :) |
23:00.20 | lesouvage | How do I disable outbound sip calls with unregistered (soft)phones. |
23:00.29 | achandra | also check /etc/reslov.conf for dns entry as well as /etc/host file |
23:01.14 | binhex | achandra: NETWORKING=yes |
23:01.14 | binhex | HOSTNAME=uioipbx.ioint.com |
23:04.23 | binhex | achandra: /etc/reslov.conf is enpty and /etc/host has localhost and my sip provider |
23:05.23 | Dr-Linux | quick question, my PRI T1 provider is my US datacenter, what framing and coding i should use in zaptel.conf? |
23:05.26 | dlynes_office | [TK]D-Fender: ah...it's not that remote server you were talking about, then, with the a200 units in it |
23:05.29 | Dr-Linux | span=1,1,0,esf,b8zs << is fine? |
23:05.42 | dlynes_office | Dr-Linux: ask your telco |
23:05.49 | drray | it depends on your telco |
23:05.51 | dlynes_office | Dr-Linux: they should've told you what it was when they hooked it up |
23:06.17 | drray | your telco can frame it anyway you want, but you can as well |
23:06.44 | Dr-Linux | i see |
23:06.57 | Dr-Linux | so any signalling will be no issues with asterisk? |
23:07.08 | drray | well, any of the big ones |
23:07.21 | achandra | binhex: what are you using for dns resolution..dont you need resolv.conf |
23:07.48 | Dr-Linux | actually we are already using few T1 lines from the same datacenter provider, but that's not asterisk, that's in TV |
23:08.02 | Dr-Linux | drray: didn't understand your last clue? |
23:08.29 | [TK]D-Fender | dlynes_office : No, I don't care about them, this is for ME :) |
23:08.36 | drray | I doubt a telco is going to give you a t1/pri that asterisk can't handle |
23:08.39 | achandra | binhex: also can you let me know what the ifcfg-eth0 file has in it?? |
23:08.47 | [TK]D-Fender | dlynes_office : I have 1 linux box used a my "everything" box, and run pure VoIP here |
23:09.15 | binhex | achandra: search ioint.com |
23:09.15 | binhex | nameserver 192.168.167.3 |
23:09.15 | binhex | nameserver 192.168.167.4 |
23:09.33 | binhex | sorry fat figered pico :) |
23:09.43 | dlynes_office | [TK]D-Fender: so selfish :) |
23:10.51 | achandra | binhex: two internal dns servers that looks okay |
23:10.51 | achandra | binhex: and that ifcfg-eth0 file? |
23:10.51 | terrapen | damned extensions...for some reason, my pattern match for _011. works but _9011. does not |
23:10.52 | terrapen | it actually matches the pattern for _9011. but before I can finish dialing, the polycom sends the call |
23:10.53 | drray | ignore pat? |
23:11.02 | terrapen | even though I have it configured not to send until the user presses send |
23:11.14 | terrapen | drray: ? |
23:11.30 | [TK]D-Fender | terrapen : You need to make sure your polycom dialplan matches your * dialplan. |
23:11.41 | terrapen | d-fender...k, i'll check that out |
23:11.52 | [TK]D-Fender | terrapen : paste it here when in doubt |
23:12.16 | terrapen | d-fender, i want the polycom to send to asterisk no matter what. i dont want it to do its own dialplan |
23:12.24 | terrapen | i'll pastebin my dialplan for ya |
23:12.42 | [TK]D-Fender | terrapen : So maye it appropriately generic. tahts what I do. |
23:13.11 | [TK]D-Fender | terrapen : #.T|X.T|*.T |
23:13.37 | [TK]D-Fender | terrapen : thats it. EVERYTHING requires Send/Timeout/# to terminate. |
23:14.03 | [TK]D-Fender | terrapen : healthy that way. I have ditched the concept of dialing 9 to dial out. its antiquated. |
23:14.50 | terrapen | well, our asterisk system runs in conjunction with two legacy PBXes, which use 9 |
23:15.06 | terrapen | so i want to keep it standardized because my users are...uh....stuck in their ways |
23:15.22 | achandra | binhex: any luck on that ifcfg-eth0 file? |
23:15.32 | achandra | or are you bonding interfaces? |
23:15.43 | [TK]D-Fender | terrapen : How unfortunate... |
23:15.55 | terrapen | http://pastebin.com/736104 |
23:16.00 | terrapen | d-fender, tell me about it... |
23:16.44 | *** join/#asterisk binhex (n=bob@216.31.167.125) |
23:17.16 | terrapen | for 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-Fender | terrapen [19:13] <[TK]D-Fender> terrapen : #.T|X.T|*.T |
23:17.34 | terrapen | strangely enought, _011. matches fine |
23:17.49 | terrapen | what does that mean? is that a dialplan regex of some kind? |
23:18.00 | [TK]D-Fender | terrapen : not for *, its for your Polycom's... |
23:18.05 | [TK]D-Fender | terrapen : 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) |
23:18.13 | terrapen | hrmmm |
23:18.25 | [TK]D-Fender | ist the Polycom that needs to relearn |
23:18.52 | binhex | achandra : I got booted for sending the file |
23:19.36 | terrapen | looks like maybe I need to monkey with <digitmap/> |
23:19.44 | terrapen | because it has: 9]xxxxxxxxx| |
23:19.52 | terrapen | err |
23:20.00 | [TK]D-Fender | terrapen : Thats what I've been saying since the start..... |
23:20.20 | terrapen | sorry, I was confused about what you're talking about |
23:20.27 | terrapen | lemme try that |
23:20.33 | *** join/#asterisk RoyK (n=roy@28.80-203-106.nextgentel.com) |
23:20.38 | achandra | binhex: use pastebin |
23:21.17 | terrapen | d-fender, sorry...i do appreciate your help...i was just confused. i know nothing about polycom dialplans |
23:22.21 | achandra | binhex: see private message |
23:22.45 | docelm0 | Im about to do the unthinkable.. Im gonna install A@H!!!!! |
23:23.02 | terrapen | sooooo slow |
23:23.15 | terrapen | docelm0: wimp |
23:23.36 | docelm0 | terrapen if you knew who you were talking to wimp wouldnt come out of yer mouth |
23:23.48 | docelm0 | Would someone clue him in on who I am please? |
23:23.59 | sevard | Who has PRI and can I see your /etc/zaptel.conf and /etc/asterisk/zapata.conf |
23:24.42 | [TK]D-Fender | Thats pre-emptive produce! For sins to be commited! |
23:24.46 | terrapen | i thot you were serious. |
23:25.12 | terrapen | besides, 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) |
23:25.30 | RoyK | zoa: 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) |
23:25.40 | docelm0 | sup tk |
23:25.42 | gmfm | sevard: i'll put mine up in a sec |
23:25.49 | docelm0 | And I am serious.. Im too lazy to download and install linux |
23:25.50 | [TK]D-Fender | heh, docelm0, ntm |
23:25.54 | *** part/#asterisk AlexCTI (n=alex@adsl-074-238-025-003.sip.mia.bellsouth.net) |
23:25.54 | sevard | gmfm: are you in the united states? |
23:25.55 | archimedes_xyz | evening all.. anybody having problems resolving DNS for cvs.digium.com? |
23:25.57 | gmfm | yes |
23:26.06 | sevard | gmfm: what kind of setup do you have if you don't mind me asking |
23:26.15 | docelm0 | And I need an asterisk box @ my house like tomorrow for my support lines.. |
23:26.21 | sevard | I think I have everything figured out I just would really feel comfortable with a matching config :) |
23:26.22 | terrapen | i generally distrust people who say, "if you knew who i was..." |
23:26.25 | terrapen | :P |
23:26.41 | [TK]D-Fender | docelm0 : Too lazy? That IS lame.... I do that at work (or as some would refer to it : INTEAD OF) |
23:26.54 | docelm0 | nah dude.. been in the scene for 2.5 years, dCAP certified, and run one of the largest asterisk installs in the US |
23:27.00 | terrapen | cool. |
23:27.06 | [TK]D-Fender | terrapen : The statement is kinda self-defeating, isn't it :) |
23:27.10 | terrapen | well, i'll be catching up with ya :) |
23:27.17 | docelm0 | ya ya ya.. |
23:27.17 | terrapen | how many phones do you have? |
23:27.18 | gmfm | sevard: * 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 |
23:27.25 | docelm0 | me? |
23:27.47 | RoyK | <PROTECTED> |
23:27.51 | terrapen | d-fender, yeah, you often hear the other version, "If you know who my [dad/family/lawyer] was..." |
23:27.55 | terrapen | docelm0, yeah |
23:27.56 | docelm0 | ZOA! |
23:28.27 | [TK]D-Fender | Can't wait to trade up my IP 301 for an IP 430 :) |
23:28.29 | docelm0 | which 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 |
23:28.37 | docelm0 | so to count phones really doesnt mean anything |
23:28.42 | sevard | gmfm: so you have one PRI going into a TDM card? what model? |
23:28.58 | terrapen | docelm0, ok, so how big? |
23:29.11 | terrapen | ok |
23:29.20 | [TK]D-Fender | terrapen : My receptionist has that.... I'm jsut wanting the IP 430 for my HOME.... |
23:29.29 | docelm0 | What do you mean how big? |
23:29.36 | terrapen | we're not that big. I will have 400 phones in a few months |
23:29.43 | terrapen | mostly in a call center |
23:29.59 | gmfm | sevard: 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 |
23:30.05 | Zodiacal | is 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... |
23:30.11 | docelm0 | Where's yer call center? |
23:30.13 | terrapen | no 2M daily or anything...but we're a retailer :P |
23:30.18 | docelm0 | brb.. need to kick something |
23:30.21 | *** join/#asterisk dlynes_office (n=dlynes@216.251.149.66) |
23:30.26 | terrapen | docelm0, salt lake city and very shortly. all over the US |
23:30.35 | gmfm | http://pastebin.com/736130 |
23:30.41 | sevard | gmfm: so you're doing PRI -> * -> PRI, not PRI -> * -> SIP |
23:30.45 | gmfm | ^^sevard |
23:30.49 | znoG | hey, does Asterisk support 484 (incomplete address response) in SIP? |
23:31.18 | archimedes_xyz | evening all.. anybody having problems resolving DNS for cvs.digium.com? |
23:31.22 | terrapen | d-fender, that works like a champ, THANK YOU |
23:31.27 | terrapen | you are the man. |
23:31.45 | dlynes_office | archimedes_xyz: that would appear to be the case, yes |
23:32.02 | CunningPike | archimedes_xyz: Read the list - CVS is being decommissioned |
23:32.07 | dlynes_office | archimedes_xyz: but that is what's supposed to happen |
23:32.16 | archimedes_xyz | dlynes, thanks just wanted to make sure it's not just me. |
23:32.17 | dlynes_office | archimedes_xyz: cvs is no more...it's been decommissioned |
23:32.29 | gmfm | sevard: 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) |
23:32.34 | dlynes_office | oops...didn't see cp was alive there |
23:32.39 | *** join/#asterisk achandra (n=achandra@12.44.122.130) |
23:32.41 | sevard | gmfm: facilityenable=yes ? |
23:33.18 | sevard | gmfm: also I must have missed [trunkgroups] didn't mention that in the wiki |
23:33.27 | pjchilds | znoG, looks like 484 has some mention in chan_sip.c |
23:33.43 | gmfm | sevard: 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. |
23:34.03 | archimedes_xyz | dlynes, what list? |
23:34.08 | dlynes_office | it's like cunningpike just comes out of the woodwork somewhere |
23:34.15 | dlynes_office | archimedes_xyz: asterisk-announcements |
23:34.29 | CunningPike | dlynes_office: I decloak |
23:34.33 | znoG | pjchilds: nice, gonna try the early dial stuff for the Grandstream then :) |
23:34.42 | dlynes_office | archimedes_xyz: i would imagine the asterisk homepage, too |
23:34.54 | sevard | gmfm: why don't you have usercallerid=yes and callerid=asreceived |
23:34.59 | achandra | pjchilds: thanks for your help yesterday on openser lb... the new 1.0.1 deals with failover just fine.. :) |
23:35.00 | gmfm | sevard: [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) |
23:35.02 | dlynes_office | archimedes_xyz: i'm really surprised you got caught though...svn's been the standard for i don't know how long |
23:35.35 | pjchilds | achandra, neat -- it would be interesting to see you ser.cfg :) |
23:35.59 | terrapen | man, i gotta get out of here |
23:36.03 | terrapen | it's mountain biking time!!! |
23:37.05 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
23:37.14 | gmfm | sevard: 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) |
23:37.39 | docelm0 | terra I dont necessarily define size of an intall on how many phones you have but more how many concurrent calls you push |
23:37.49 | terrapen | d-fender, thanks again man. that rocks |
23:37.57 | terrapen | docelm0, absolutely |
23:38.18 | [TK]D-Fender | terrapen : ywc |
23:38.42 | dlynes_office | [TK]D-Fender: ywc? your wife's cute? |
23:38.45 | docelm0 | Which in my case I push on adverage 200+ channels at a slow time and upwards of 800 at peak |
23:38.57 | docelm0 | s/channels/calls |
23:39.07 | docelm0 | dumb bot |
23:39.29 | dlynes_office | dumb pot |
23:39.32 | [TK]D-Fender | docelm0 : you need to double // it like tihs |
23:39.34 | dlynes_office | s/pot/bot/ |
23:39.45 | [TK]D-Fender | s/tihs/this/ |
23:40.15 | docelm0 | hmm didnt work in mirc |
23:40.17 | dlynes_office | dumb docelm0 :) |
23:40.20 | [TK]D-Fender | dlynes_office : You're Welcome |
23:40.24 | docelm0 | soemthing like that |
23:40.27 | docelm0 | something |
23:40.38 | Dr-Linux | // |
23:40.44 | pjchilds | ~spank pjchilds |
23:40.45 | jbot | ACTION bends pjchilds over his knee and tatoos 'ibot' on pjchilds's pasty white buttocks. |
23:41.06 | terrapen | docelm0, we'll probably only push 3 PRI's full at a time |
23:41.22 | terrapen | maybe 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) |
23:42.13 | Dr-Linux | dlynes_office: today we ordered a new DELL PE 2850 server. |
23:42.19 | chino | hi |
23:42.26 | terrapen | asterisk is currently being used as a bridge between two crappy old legacy PBXes here, which is kind of neat |
23:42.28 | dlynes_office | oooh |
23:42.41 | dlynes_office | Dr-Linux: congratulations on figuring out how to use a web order form :) |
23:42.43 | chino | do i need to configure dial plans if im not using fxo or fxs ports ? |
23:42.45 | sevard | gmfm: When you get a PRI from a telco is it always CPE? Or could sometimes it be pri_net ? |
23:43.21 | [TK]D-Fender | Dr-Linux : So why the new server? Growing that fast now? |
23:43.40 | [TK]D-Fender | sevard : in any sane scenario they'll play net... |
23:43.43 | gmfm | sevard: you will always be CPE to a telco, unless you come from a backwards town |
23:43.49 | Dr-Linux | dlynes_office: thanks, but this is not our first server, we have already alot of Sun and Dell PE's |
23:44.05 | chino | how do i tell both endpoints to communicate with one another without relaying off the server? |
23:44.12 | sevard | gmfm: are you right or is [TK]D-Fender right? :) |
23:44.16 | dlynes_office | Dr-Linux: but why is the pe 2850 such a good thing? |
23:44.23 | Dr-Linux | [TK]D-Fender: we are placing the new server at datacenter, |
23:44.41 | docelm0 | Do you guys like the 2850? Have you messed with the 1850's much? |
23:44.45 | Dr-Linux | dlynes_office: for IVR solutions |
23:44.47 | gmfm | sevard: we effectively said the same thing... the telco is NET, you are CPE |
23:45.05 | sevard | Heh. Alright. |
23:45.07 | znoG | pjchilds: yeah, just had a look at chan_sip.c ... looks like it supports it |
23:45.08 | Dr-Linux | 2850 looks good |
23:45.27 | docelm0 | I bought 10 1850's the blade system they have I am very unimpressed |
23:45.30 | sevard | Well then, I think I have everything right. I just need to wait till 7:00 A.M. when they flip the switch |
23:45.35 | dlynes_office | Dr-Linux: yeah...we always build our own servers |
23:45.41 | gmfm | sevard: when all else fails... play with it until it works :-) |
23:45.44 | dlynes_office | Dr-Linux: so i wouldn't even know what a 2850 is |
23:45.47 | sevard | You don't need language=en on a PRI? |
23:45.50 | dlynes_office | Dr-Linux: other than that it's a rackmount |
23:46.28 | sevard | oops, yes you do :| |
23:46.33 | gmfm | sevard: 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 |
23:46.39 | Dr-Linux | dlynes_office: before ordering the server, i came here and ask for suggestion, but didn't get any answer |
23:46.49 | sevard | gmfm: ahh. I see. |
23:47.16 | dlynes_office | Dr-Linux: yeah...nobody could probably recommend anything to you besides brand name anyways, because nobody's local to you |
23:48.10 | gmfm | sevard: 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 |
23:48.35 | Dr-Linux | dlynes_office: nope, we are purchasing in the US, not here, |
23:49.05 | sevard | gmfm: I got unknown signalling method pri_cpe |
23:49.09 | Dr-Linux | i 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) |
23:49.38 | dlynes_office | Dr-Linux: another server that probably would have worked well is a Sun Cobalt box |
23:49.39 | sevard | gmfm: |
23:49.45 | ManxPower | anyone having a sound file of Allison saying "If your are sending a fax, please press "start" now."? |
23:49.51 | Dr-Linux | hhm.. |
23:50.10 | gmfm | sevard: that's odd |
23:50.19 | dlynes_office | Dr-Linux: but of course they wouldn't recommend that because it runs AMD processors, not Sun processors :) |
23:50.39 | sevard | gmfm: tell me about it :\ |
23:50.48 | Dr-Linux | dlynes_office: first we were interesting in X4100 from Sun, but that was looking low profile |
23:51.14 | dlynes_office | Dr-Linux: the cobalt boxes are their low end Linux boxes |
23:51.25 | gmfm | sevard: 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 |
23:51.39 | zparta | woohoo found a place on the net that sells voip stuff that i trust |
23:51.41 | Dr-Linux | dlynes_office: currently we are using 2 brands DELL and Sun, for all our production servers, and PBX .. but those are not asterisk |
23:51.58 | gmfm | sevard: oh hey did you run ztcfg -vv after setting up zaptel.conf? |
23:52.03 | sevard | gmfm: 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) |
23:52.29 | sevard | gmfm: yeah |
23:52.33 | loonacy | I'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. |
23:52.43 | Dr-Linux | dlynes_office: do you think DELL 2850 is a good choice? for 2 TE210P .. ? |
23:53.04 | gmfm | sevard: did you build/install libpri before asterisk? |
23:53.09 | dlynes_office | Dr-Linux: no idea, but any of their SunFire servers would be fine |
23:53.18 | [TK]D-Fender | loonacy : use AMI to connect to * and issue "show database" and parse it. |
23:53.30 | sevard | gmfm: 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-Fender | loonacy : Or use DB1 raw on the * DB file direct. |
23:53.53 | dlynes_office | Dr-Linux: i've only got one dell machine, and it was only cause i got a good deal on it from ebay |
23:54.00 | gmfm | sevard: libpri won't necessarily affect a TDM400P because that's not a T1/PRI card |
23:54.23 | sevard | gmfm: That's good to know. I'll look furthure |
23:55.21 | gmfm | sevard: best of luck on the turn-up tomorrow... i get to go home now :-) |
23:55.35 | sevard | gmfm: have fun |
23:55.39 | sevard | gmfm: thanks man |
23:56.42 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
23:57.15 | Dr-Linux | dlynes_office: this server will have only IVR's in AGI.. caller will do everything via IVR. |
23:57.54 | sevard | oh crap, now where is libpri |
23:58.08 | Dr-Linux | sevard? |
23:58.14 | jayk- | libpri? |
23:58.17 | jayk- | it should be in /usr/lib? |
23:58.38 | sevard | I don't have libpri |
23:58.40 | Dr-Linux | you need to compile libpri first |
23:58.44 | jayk- | you can get it from www.asterisk.org |
23:59.08 | sevard | I thought it should have came in the zaptel driver archive |
23:59.26 | jayk- | seperate |
23:59.34 | Dr-Linux | sevard: if you are using PRI, then you need libpri, becore asterisk |
23:59.37 | chino | can you make your voip server open or do you have to add a user section for every user in the system ? |
23:59.42 | jayk- | you are installing from package? |
23:59.49 | chino | to the sip.conf... |
23:59.51 | sevard | Dr-Linux: I understand that, where is it? |