| 00:00.10 | dijungal | i'm beginning to think sp |
| 00:00.12 | dijungal | so |
| 00:00.38 | *** join/#asterisk SwK (n=SwK@user-69-73-37-99.knology.net) |
| 00:00.48 | JT | h.232? |
| 00:01.45 | *** join/#asterisk Infested (n=infested@24.148.112.10) |
| 00:01.47 | dijungal | i just tried "h323 set debug" |
| 00:02.01 | dijungal | and i'm callin in |
| 00:02.20 | dijungal | same thing asteris just sits there |
| 00:02.37 | JT | i assume you've at least done some packet dumping |
| 00:02.45 | dijungal | yes i have |
| 00:02.57 | CrashHD | is 603 a normal sip return code for a busy line? |
| 00:04.55 | *** join/#asterisk Strom_M (n=strom@60-241-200-26.static.tpgi.com.au) |
| 00:05.47 | *** join/#asterisk xpander4 (n=gaston@72.242.62.90) |
| 00:10.45 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 00:10.52 | _VoiceMeUp_COM | sorry was disc |
| 00:10.57 | dijungal | http://pastebin.ca/580160 |
| 00:11.07 | dijungal | i've added the tcpdump info |
| 00:11.27 | JT | is asterisk listening on the H.323 port? |
| 00:11.55 | dijungal | yes |
| 00:13.08 | dijungal | but just to check how would u test to see if it is? |
| 00:13.13 | JT | probably won't solve your problem, but this is a very bad idea: |
| 00:13.15 | JT | # |
| 00:13.15 | JT | allow=g729 |
| 00:13.15 | JT | # |
| 00:13.15 | JT | allow=gsm |
| 00:13.17 | JT | # |
| 00:13.20 | JT | dtmfmode=inband |
| 00:13.25 | dijungal | k |
| 00:13.26 | JT | well almost none of us use H.323 |
| 00:13.32 | dijungal | what do u suggest? |
| 00:13.38 | dijungal | k |
| 00:13.38 | JT | SIP |
| 00:13.44 | dijungal | that's why it's so hard to get support |
| 00:13.57 | JT | yes |
| 00:14.00 | dijungal | i know i'm a SIP user myself.. this h.323 is beating me |
| 00:14.15 | JT | but never use inband dtmf over a compressed codec |
| 00:14.26 | dijungal | and no buying a next cisco router not on my budget |
| 00:14.31 | dijungal | k |
| 00:14.40 | dijungal | i should use rfc2833 |
| 00:16.24 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 00:16.26 | JT | dijungal: what's the ip of asterisk, and ip of the cisco? |
| 00:17.20 | dijungal | cisco -> 208.45.130.100 asterisk -> 205.244.148.44 |
| 00:17.33 | *** part/#asterisk Nuitari (n=Nuitari@142.46.207.230) |
| 00:18.08 | JT | dijungal: i take it you didn't actually READ the tcpdump output |
| 00:18.17 | JT | just nodded your head at it |
| 00:19.22 | JT | dijungal: it patently obvious at least what your first problem is from the tcpdump output |
| 00:19.35 | dijungal | tell me... |
| 00:19.59 | dijungal | "208.45.130.100 unreachable" ? |
| 00:20.09 | JT | 208.45.130.100 unreachable - admin prohibited filter for IP |
| 00:20.16 | JT | switch off the damned firewall. |
| 00:20.28 | dijungal | huh.. firewall.. hmm.. |
| 00:20.34 | dijungal | can't remember having one |
| 00:20.36 | dijungal | hold |
| 00:20.44 | JT | iptables -L |
| 00:21.23 | dijungal | that's clean |
| 00:21.25 | dijungal | no rules |
| 00:21.34 | dijungal | and i can ping that ip from the asterisk box |
| 00:21.43 | JT | ping has no revelance |
| 00:21.46 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 00:21.56 | JT | it's probably a firewall on the cisco then |
| 00:22.03 | dijungal | ohooo i thought cause it said "icmp" |
| 00:22.07 | *** join/#asterisk BadPacket (n=NoCarrie@unaffiliated/badpacket) |
| 00:22.13 | dijungal | probably |
| 00:22.27 | JT | ping only shows that pings is being permitted |
| 00:23.06 | riddlebox | can someone help me I am trying to call another asterisk server from mine, and use its ivr, I cannot press any buttons though, nothing happens? I have dtmfmode=auto set and I can call other places and it works |
| 00:23.23 | dijungal | i guess i need to open the 1720 port on the cisco.. actually i think there's an NS25 infront that cisco router.. i'll have to check |
| 00:23.35 | JT | using what protocol and codec? |
| 00:23.57 | JT | dijungal: so you don't even know what the network setup is.... useful :P |
| 00:24.18 | dijungal | lol... i'm not even in the same country |
| 00:24.26 | dijungal | not even in the same continent .. lol |
| 00:24.34 | JT | nice |
| 00:24.40 | dijungal | very :( |
| 00:24.43 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 00:25.21 | dijungal | oooh and i don't even control the cisco and firewall equipment... |
| 00:25.21 | dijungal | now i gotta go to the network guy and convince him his cisco needs reconfiguring |
| 00:25.22 | dijungal | or firewall or something |
| 00:25.39 | dijungal | i'm just making sure my asterisk end is covered |
| 00:25.53 | dijungal | thanks for the help tho :) |
| 00:26.14 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 00:26.48 | JT | it looks like you are receiving icmp message 36 back |
| 00:26.55 | JT | but i can't tell what router is giving it |
| 00:27.05 | JT | so there's still a posibility it's on your network |
| 00:28.58 | *** join/#asterisk breanna_ (n=brea@c-71-195-248-169.hsd1.ut.comcast.net) |
| 00:32.07 | diclophis-work | hello all |
| 00:32.11 | diclophis-work | how can i call in and listen to a call? |
| 00:32.12 | diclophis-work | Barge? |
| 00:33.19 | [TK]D-Fender | diclophis-work, "show application chanspy" |
| 00:33.22 | diclophis-work | cool |
| 00:33.23 | diclophis-work | thanks |
| 00:35.26 | diclophis-work | can i spy on local channels? |
| 00:45.04 | dijungal | TKD-Fenfer: is there some program i can use to scan the ip to see what ports are open? |
| 00:45.08 | [TK]D-Fender | A channel is a channel is a channel |
| 00:45.25 | [TK]D-Fender | dijungal, "netstat -an" |
| 00:45.41 | dijungal | no i mean scan the cisco router |
| 00:45.46 | dijungal | not the asterisk |
| 00:45.58 | [TK]D-Fender | dijungal, "name nmap" |
| 00:46.06 | dijungal | man nmap |
| 00:46.11 | *** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net) |
| 00:46.39 | diclophis-work | is it possible for an asterisk server to call itself over sip? |
| 00:48.15 | _VoiceMeUp_COM | JerJer you back ? |
| 00:48.22 | _VoiceMeUp_COM | diclophis-work use LOCAL |
| 00:48.29 | _VoiceMeUp_COM | LOCAL/EXTEN@CONTEXT |
| 00:48.33 | diclophis-work | ok cool |
| 00:48.34 | _VoiceMeUp_COM | \n |
| 00:48.38 | diclophis-work | but that doesnt work with ChanSpy |
| 00:48.42 | _VoiceMeUp_COM | hmmm |
| 00:48.50 | _VoiceMeUp_COM | tought those 2 where bad nayhow |
| 00:48.51 | diclophis-work | that is, i cant hear anything when i spy on two local channels connected to each other |
| 00:48.54 | *** join/#asterisk Dert1cK (n=fan@88.84.207.114) |
| 00:48.57 | Dert1cK | Here there are girls? |
| 00:49.09 | *** join/#asterisk Avochelm (n=damo@gw-morphett.koalatelecom.com.au) |
| 00:49.10 | _VoiceMeUp_COM | you could.. make a virtual meeting with ocnferences and one on mute |
| 00:49.14 | _VoiceMeUp_COM | and trasnfer parties there |
| 00:49.21 | _VoiceMeUp_COM | Dert1cK this is not a dating channel ;) |
| 00:49.26 | JT | Dert1cK: this is not a pickup channel |
| 00:49.33 | Strom_M | you want #asstricks |
| 00:49.34 | _VoiceMeUp_COM | JT LOL beat you to it |
| 00:50.08 | _VoiceMeUp_COM | show application Pickup |
| 00:50.28 | Strom_M | hahahahaha |
| 00:50.38 | Strom_M | "Hey there...I'm here to punch my tool into your connecting block..." |
| 00:50.39 | xpander4 | me /joins #asstricks ! |
| 00:51.58 | Dert1cK | i Russian!! sorry ! I here first time |
| 00:52.27 | Strom_M | Dert1cK: we talk about TELEPHONES here |
| 00:52.36 | Strom_M | what the hell possessed you to ask about women? :) |
| 00:52.57 | _VoiceMeUp_COM | ahahah Strom_M just got the annalogy |
| 00:53.12 | Strom_M | hehehe |
| 00:54.20 | JT | "I'd like to punch that down, hard" |
| 00:54.55 | Dert1cK | Strom_M » And on what channel of the girl? |
| 00:55.46 | _VoiceMeUp_COM | ? |
| 00:55.47 | _VoiceMeUp_COM | lol |
| 00:56.14 | _VoiceMeUp_COM | try #hoes or #girls |
| 00:56.18 | _VoiceMeUp_COM | dpeending on what you seek |
| 00:56.19 | Strom_M | Dert1cK: well I suppose you could call the operator and ask her for the correct time |
| 00:56.35 | Dert1cK | _VoiceMeUp_COM » Be not dared over me |
| 00:56.46 | Strom_M | "For a good time, call 555-2368" "For the correct time, call 0" |
| 00:57.29 | breanna_ | Dert1cK: Go in #freenode and ask them... and stop sending me messages |
| 00:58.03 | [TK]D-Fender | Strom_M, .....867-5309 <- |
| 00:59.35 | JT | Dert1cK: sad sad little man |
| 01:00.52 | *** join/#asterisk sid (n=unstable@tor/regular/sid) |
| 01:01.12 | _VoiceMeUp_COM | lol |
| 01:01.16 | flenders | oh god, that was funny... |
| 01:01.44 | flenders | why on earth would someone join a channel called 'asterisk' and ask for girls? |
| 01:01.45 | JT | -!- Dert1cK [n=fan@88.84.207.114] has joined #freenode |
| 01:01.54 | JT | < Dert1cK> hi all . Here there are girls? |
| 01:01.59 | _VoiceMeUp_COM | that jenny's number |
| 01:02.00 | _VoiceMeUp_COM | lol |
| 01:02.01 | _VoiceMeUp_COM | 867-5309 |
| 01:02.04 | flenders | hahahha |
| 01:02.09 | _VoiceMeUp_COM | i heard they blocke dit |
| 01:02.16 | breanna_ | flenders: dunno... but he's askign in #freenode now |
| 01:02.24 | _VoiceMeUp_COM | even people have voice reocrdings saying. hi this is jenny lol |
| 01:02.30 | flenders | well, better than in here |
| 01:02.32 | _VoiceMeUp_COM | http://en.wikipedia.org/wiki/867-5309 |
| 01:02.35 | Strom_M | _VoiceMeUp_COM: no, you can still get 867-5309 assigned if you're on an 837 exchange :) |
| 01:02.38 | Strom_M | er |
| 01:02.39 | Strom_M | 867 |
| 01:03.14 | file | we have an 867 exchange here... but I doubt the telco would assign it |
| 01:03.47 | Strom_M | file: you should request the number |
| 01:04.02 | Strom_M | hell, you know what my home number is, and I got that just by asking :) |
| 01:04.12 | file | Strom_M: I would, if the SPA3102 could figure out incoming distinctive ring |
| 01:04.31 | Strom_M | file: but but but you don't have a tdm card? :D |
| 01:04.55 | file | well, I do |
| 01:06.13 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 01:06.27 | _VoiceMeUp_COM | <PROTECTED> |
| 01:06.29 | _VoiceMeUp_COM | got this |
| 01:06.59 | *** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) |
| 01:07.16 | Strom_M | _VoiceMeUp_COM: you've got DIDs in that block? |
| 01:08.50 | _VoiceMeUp_COM | cecking |
| 01:08.51 | _VoiceMeUp_COM | yes |
| 01:08.55 | _VoiceMeUp_COM | got lots lol |
| 01:09.02 | Strom_M | 5309? :) |
| 01:10.08 | _VoiceMeUp_COM | nope |
| 01:10.14 | Strom_M | damnit |
| 01:11.00 | _VoiceMeUp_COM | <PROTECTED> |
| 01:11.02 | _VoiceMeUp_COM | lots |
| 01:11.02 | _VoiceMeUp_COM | lol |
| 01:11.21 | Strom_M | yeah, but you don't have 5309 in that prefix |
| 01:11.26 | Strom_M | so what's the point |
| 01:11.59 | _VoiceMeUp_COM | nah |
| 01:12.00 | _VoiceMeUp_COM | i know |
| 01:12.04 | _VoiceMeUp_COM | checked up to 416 |
| 01:12.06 | _VoiceMeUp_COM | but to long |
| 01:12.06 | _VoiceMeUp_COM | lol |
| 01:12.15 | _VoiceMeUp_COM | i nee da vanity tool for our stuff |
| 01:12.22 | _VoiceMeUp_COM | ill get the programmers to write the module |
| 01:12.29 | *** join/#asterisk CrashSys (n=kumba@158-211.187-72.tampabay.res.rr.com) |
| 01:13.02 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 01:13.02 | *** mode/#asterisk [+o anthm] by ChanServ |
| 01:13.26 | CrashSys | Any cable/telco guys out there used to working in a cloud? Http://www.crashsys.com/~kumba/bc3.jpg Any recommendations where to start? :D |
| 01:15.25 | _VoiceMeUp_COM | burn it all down |
| 01:15.32 | CrashSys | I like that idea |
| 01:15.36 | JT | does it help that my first thought was "that's fucked" ? |
| 01:15.47 | CrashSys | Let the insurance pay me to strip it building and re-run :D |
| 01:15.57 | *** join/#asterisk phix (i=threat@60-240-43-214.static.tpgi.com.au) |
| 01:16.13 | CrashSys | http://www.crashsys.com/~kumba/bc2.jpg (Evidently this is an approved wiring hanging method too) |
| 01:16.15 | _VoiceMeUp_COM | hheehe |
| 01:16.21 | macTijn | oh shit. |
| 01:16.26 | CrashSys | check out bc1.jpg and bc4.jpg for more fun |
| 01:16.32 | _VoiceMeUp_COM | listen dude , you can pay men 150$ an hour to untangle.. total cost 50k |
| 01:16.35 | macTijn | where is that ? |
| 01:16.36 | _VoiceMeUp_COM | or burn the place down |
| 01:16.38 | _VoiceMeUp_COM | ;) |
| 01:16.40 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 01:17.06 | macTijn | it scares me :( |
| 01:17.09 | CrashSys | A job where the guy has me working up a quote to bring the 200-pairs of demarc to a wiring closet and rewire the building with a kind of mesh-setup... |
| 01:17.19 | macTijn | looks worse than my cage at level3 |
| 01:17.25 | CrashSys | there's only like 3 tenants in the building at the moment so there's not much active pairs to worry about... |
| 01:17.43 | macTijn | CrashSys: it's the active ones you have to worry about ;) |
| 01:17.48 | CrashSys | That bc4.jpg box has like 5 25-pair cables terminating inside it... cant even see the binding posts/etc... |
| 01:17.58 | Strom_M | oh, well there's your problem: |
| 01:18.00 | Strom_M | G |
| 01:18.01 | Strom_M | T |
| 01:18.02 | Strom_M | E |
| 01:18.06 | CrashSys | macTijn: Yeah... be nice to tell them they will be without for a week, and go have a vacation! |
| 01:18.21 | Strom_M | Guaranteed Trouble Everytime |
| 01:18.25 | Strom_M | Get Telephone Elsewhere |
| 01:18.26 | _VoiceMeUp_COM | ahahaha |
| 01:18.26 | CrashSys | Strom: d00d, GTE hasn't existed in florida for like 6 years... it's verizon now... and that is the ONLY labeled pair in that box :) |
| 01:18.27 | _VoiceMeUp_COM | GT ? |
| 01:18.31 | Strom_M | the Great Telephone Experiment |
| 01:18.35 | _VoiceMeUp_COM | lol |
| 01:18.38 | _VoiceMeUp_COM | bout primus |
| 01:18.43 | Strom_M | CrashSys: it's still GTE, just with a different name :) |
| 01:18.43 | _VoiceMeUp_COM | make on on theyr ass |
| 01:18.56 | CrashSys | Pretty much |
| 01:18.57 | _VoiceMeUp_COM | coz primus sucks |
| 01:19.06 | CrashSys | But gives you an idea of how long ago that box was popped... |
| 01:19.16 | macTijn | Goddamned Telephony Engineers. |
| 01:19.36 | macTijn | ;) |
| 01:19.42 | CrashSys | That is obviously verizon's demarc for the building... |
| 01:19.48 | _VoiceMeUp_COM | Good Till Ends |
| 01:19.49 | macTijn | anyway |
| 01:19.55 | macTijn | off to bed with me |
| 01:19.56 | macTijn | nn all |
| 01:19.57 | CrashSys | I cant believe what cable installer ran them that way... |
| 01:20.09 | CrashSys | it's not even OSP cable... regular PVC Cat3 |
| 01:20.11 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 01:21.43 | *** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk) |
| 01:22.08 | CrashSys | Anyways, that is what happens when low-voltage comm installers are not required to be licensed :D |
| 01:25.20 | *** join/#asterisk logyati (n=paulo@201.29.18.64) |
| 01:32.49 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-67-180-78-55.hsd1.ca.comcast.net) |
| 01:33.35 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 01:38.01 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 01:42.23 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 01:42.47 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 01:42.57 | flenders | hey, has anyone here used those polycom conference bridges? |
| 01:43.21 | flenders | like the soudstation EX? |
| 01:45.38 | Strom_M | that's not a conference bridge |
| 01:45.41 | Strom_M | that's a conference phone |
| 01:46.02 | CrashSys | Like an IP 4000 or whatever it is |
| 01:50.39 | flenders | Strom_M: and is it any good? |
| 01:51.20 | Strom_M | polycom makes the cream of the crop of conference phones :) |
| 01:51.21 | flenders | I don't think I know the difference between conference bridge and phone |
| 01:51.27 | Strom_M | a phone is a phone |
| 01:51.38 | Strom_M | a bridge is something within a telephone switch |
| 01:51.42 | flenders | let me guess, a bridge is a bridge |
| 01:51.47 | flenders | ok |
| 01:51.50 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 01:57.09 | *** part/#asterisk sid (n=unstable@tor/regular/sid) |
| 02:00.14 | CrashHD | how is everyone tonight? |
| 02:03.12 | *** join/#asterisk CrashHD (n=timf@70.96.98.65) |
| 02:16.07 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 02:22.08 | *** join/#asterisk ZaVoid (n=zavoid@c-67-165-25-195.hsd1.ct.comcast.net) |
| 02:24.06 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 02:24.06 | *** mode/#asterisk [+o mog] by ChanServ |
| 02:24.54 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 02:25.19 | CrashHD | dum de dum |
| 02:25.26 | *** join/#asterisk Kaycut (n=nada@host171.190-31-153.telecom.net.ar) |
| 02:25.38 | *** join/#asterisk _DAW (n=chatzill@adsl-241-94-68.msy.bellsouth.net) |
| 02:25.47 | Kaycut | some help is needed |
| 02:25.49 | Kaycut | hi |
| 02:26.24 | Kaycut | hello anyone |
| 02:26.34 | _DAW | hi there |
| 02:26.44 | CrashHD | have to ask a question before anyone can answer it? |
| 02:31.08 | Kaycut | yes |
| 02:31.10 | Kaycut | sorry |
| 02:31.14 | Kaycut | i was outside |
| 02:31.25 | Strom_M | no no, don't ask a question...let us GUESS what your problem is :) |
| 02:31.49 | Kaycut | i want to make an small enterprice providing voip service to customers |
| 02:31.58 | Kaycut | im in a small city |
| 02:32.30 | Kaycut | i want to give anyone a voip phone and they can make calls togheter |
| 02:32.54 | Kaycut | i dont want to use analog lines, just internet and just ip calls |
| 02:33.07 | Kaycut | can i make this with asterisk? |
| 02:33.13 | Strom_M | yes |
| 02:33.31 | Kaycut | do i need a voip service provider to do this? |
| 02:33.44 | [TK]D-Fender | Kaycut, You just said you wanted to BECOME one... |
| 02:33.46 | Kaycut | i speak spanish, and more or less english |
| 02:33.58 | [TK]D-Fender | (50$ on less) |
| 02:34.07 | Kaycut | of course |
| 02:34.09 | [TK]D-Fender | ;) |
| 02:34.26 | Kaycut | i want to become a voip provider but for free |
| 02:34.27 | [TK]D-Fender | Kaycut, no, you do not need any other service providers unless you DESIRE them |
| 02:34.45 | [TK]D-Fender | Kaycut, Then certainly you can be the central server for your little community. |
| 02:34.53 | Kaycut | ok, i dont want to use terrain lines, do i explain? |
| 02:34.59 | Kaycut | yes |
| 02:35.02 | Kaycut | for example |
| 02:35.23 | Kaycut | you have a ip phone and my neighbor another ip phne |
| 02:35.25 | JT | land lines |
| 02:35.44 | Kaycut | you pick ip your phone and call him with a number that i give |
| 02:35.57 | [TK]D-Fender | Kaycut, Yes, you can have it so everyone can call each other with them for free. |
| 02:36.04 | Kaycut | yes |
| 02:36.07 | [TK]D-Fender | Kaycut, exactly |
| 02:36.11 | Kaycut | just with a internet conection |
| 02:36.16 | Kaycut | can i do that? |
| 02:36.23 | [TK]D-Fender | Kaycut, Yes, easily |
| 02:36.29 | Kaycut | yeah |
| 02:36.34 | Kaycut | im smiling |
| 02:36.44 | Kaycut | ok |
| 02:36.50 | Kaycut | im new in this |
| 02:36.54 | Kaycut | newbie one |
| 02:36.58 | ZaVoid | anyone use the h323chan for inbound calls? |
| 02:37.04 | Strom_M | i'd recommend you hire a consultant :) |
| 02:37.06 | [TK]D-Fender | Kaycut, here : |
| 02:37.07 | [TK]D-Fender | ~book |
| 02:37.09 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 02:37.19 | [TK]D-Fender | Kaycut, Start with.. THE BOOK. Then move on to the WIKI |
| 02:37.22 | [TK]D-Fender | ~wikis |
| 02:37.23 | jbot | rumour has it, wikis is http://www.voip-info.org |
| 02:37.28 | Kaycut | zavoid, i dont understand that |
| 02:37.34 | [TK]D-Fender | Strom_M, He never said he was in a RUSH. |
| 02:37.45 | Kaycut | ok wait |
| 02:37.48 | Kaycut | slowly |
| 02:38.08 | Kaycut | waht about <ZaVoid> anyone use the h323chan for inbound calls? |
| 02:38.09 | [TK]D-Fender | Kaycut, Go downlaod that free PDF book. that should get you through a lot of the learning process. |
| 02:38.18 | Kaycut | ok |
| 02:38.22 | Kaycut | basically |
| 02:38.29 | [TK]D-Fender | Kaycut, then when you want to learn some more on some specific parts of Asterisk, heck out the WIKI. |
| 02:38.32 | Kaycut | i have the book |
| 02:39.03 | Kaycut | i want to know two thinks |
| 02:39.31 | Kaycut | one: do i really need a land line to make calls betwen my customers using ip phones? |
| 02:39.40 | Kaycut | i guess that no |
| 02:39.59 | JT | no, why would you? |
| 02:40.16 | ZaVoid | or anyone use yate? |
| 02:40.40 | JT | ZaVoid: wrong channel perhaps? :) |
| 02:40.53 | ZaVoid | nah :) |
| 02:41.09 | ZaVoid | i'd use yate to convert h.323 to sip and then send to my asterisk boxes i think |
| 02:41.17 | JT | might work |
| 02:41.24 | ZaVoid | lol maybe |
| 02:41.27 | JT | since asterisk is hopeless with H.323 |
| 02:41.45 | ZaVoid | yeah and i need clients to be able to register via h.323 too :( user/pin blah blah |
| 02:41.59 | *** join/#asterisk Kaycut (n=nada@host171.190-31-153.telecom.net.ar) |
| 02:42.04 | Kaycut | sorry |
| 02:42.44 | Kaycut | i speak spanish and a little bit of ensglish |
| 02:42.54 | Kaycut | that why im asking you zavoid |
| 02:42.55 | [TK]D-Fender | Kaycut, We got that part already... |
| 02:43.03 | Kaycut | yes |
| 02:43.09 | Kaycut | i know |
| 02:43.12 | Kaycut | sorry |
| 02:43.15 | ZaVoid | what are you asking me Kaycut ? |
| 02:43.17 | [TK]D-Fender | Kaycut, and I don't think he is talking to YOU... |
| 02:43.21 | Kaycut | ok |
| 02:43.27 | Kaycut | sorry |
| 02:43.33 | ZaVoid | i could talk to you |
| 02:43.38 | ZaVoid | hi Kaycut how ya doing? |
| 02:43.42 | Kaycut | hi |
| 02:43.56 | [TK]D-Fender | Kaycut, No, you don't need a land-line. You can just have it so they can call each other. No special hardware or any lines needed for that. |
| 02:44.11 | [TK]D-Fender | Kaycut, Just put an IP phone at each place and add them to your server. |
| 02:44.15 | Kaycut | can i become a voip service providers without a land line just for ip lines clients |
| 02:44.22 | [TK]D-Fender | Kaycut, YES |
| 02:46.18 | JT | Kaycut: stop asking the same question over and over, it's really giving us the shits |
| 02:46.28 | ZaVoid | lol |
| 02:46.29 | *** part/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 02:46.38 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 02:46.40 | [TK]D-Fender | oops |
| 02:46.51 | ZaVoid | i'm gonna lose my mind in lrqs.. back in a few |
| 02:48.24 | *** join/#asterisk ^Rocket^ (n=rocket@c-71-235-48-164.hsd1.ct.comcast.net) |
| 02:49.27 | *** join/#asterisk Kaycut (n=nada@host171.190-31-153.telecom.net.ar) |
| 02:49.33 | Kaycut | sorry again |
| 02:49.44 | Kaycut | how big the server have to be? |
| 02:49.57 | Kaycut | how fast mi internet connection have to be? |
| 02:50.31 | JT | how about READING THE BOOK? |
| 02:50.34 | Kaycut | ok\ |
| 02:50.35 | JT | and the wiki |
| 02:50.38 | [TK]D-Fender | Kaycut, Depends on how many simultaneous calls, what codecs, etc. You need to stop for a bit and do some reading. |
| 02:50.44 | Kaycut | im going there now |
| 02:50.52 | ^Rocket^ | [TK]D-Fender Hey |
| 02:51.04 | Kaycut | thanks a lot to everyone |
| 02:52.12 | CrashHD | any documentation available on the new call parking variable? |
| 02:56.46 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 03:01.42 | ^Rocket^ | How do I set asterisk to listen for SIP on prt 5060? I just installed a fresh server and asterisk, and 5060 doesn't seem to be open? |
| 03:02.06 | [TK]D-Fender | ^Rocket^, "netstat -an" |
| 03:02.26 | ^Rocket^ | ok |
| 03:02.54 | russellb | um, if chan_sip is loaded, it listens on 5060 by default (UDP only) |
| 03:03.29 | Strom_M | russellb: I have a mega-nub question: what is the default password for the aadk again? :) |
| 03:04.16 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:04.34 | ^Rocket^ | russellb chan_sip shows a 0 in the right most column when I do a "show modules" in the console |
| 03:04.59 | russellb | that's normal |
| 03:05.00 | ^Rocket^ | but "load chan_sip" says it already exists |
| 03:05.04 | ^Rocket^ | ok |
| 03:05.23 | ^Rocket^ | so with my fresh install, should I configure a sip device next? |
| 03:05.28 | russellb | 0 just means the use count ... you can't unload it when it is non-zero |
| 03:05.40 | ^Rocket^ | what's the easiest next baby step to take in setting my system up? |
| 03:06.13 | russellb | depends what you're trying to do :) |
| 03:06.38 | russellb | reading the book or wiki on the topic you're interested in is where most people go |
| 03:06.43 | russellb | ~thebook |
| 03:06.44 | jbot | it has been said that thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 03:06.57 | ^Rocket^ | for now, setup a sip phone, I already got a config for connecting to vonage softphone service |
| 03:07.11 | ^Rocket^ | I have the Oreilly book |
| 03:07.28 | ^Rocket^ | I reread a chapter the other night |
| 03:07.36 | ^Rocket^ | pages 50-100 |
| 03:08.20 | [TK]D-Fender | ^Rocket^, just keep reading. You're nowhere until you've gone through Chapter 5 at least twice :) |
| 03:08.57 | ^Rocket^ | I had a setup a year ago for a telesip account I had for a few months |
| 03:09.17 | ^Rocket^ | it's hard to see the big picture |
| 03:09.41 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 03:10.18 | *** join/#asterisk MrTelephone (n=test@bas13-toronto63-1177851021.dsl.bell.ca) |
| 03:11.29 | MrTelephone | I was having all these overrun issues with my card. Turns out you HAVE to use the telcos t1 timing if you hook up a channel bank to the second port on the dual t1 a102d card |
| 03:11.43 | [TK]D-Fender | ^Rocket^, Configure your sip phone so it can connect. Setup a basic dialplan so you can prove yuo did it right. Make some extens to dial your phone devices, access VM, etc. Then add your ITSP. Make extens to let you dial out and handle calls coming in. |
| 03:11.56 | MrTelephone | fender, do you mess with vlans alot? |
| 03:12.16 | [TK]D-Fender | MrTelephone, nope, not at all |
| 03:12.19 | MrTelephone | can the computer behind the polycom 501 be on a different vlan? |
| 03:12.25 | ZaVoid | anyone using 1.4.5? |
| 03:12.52 | MrTelephone | 1.2.12 is hard enough to get working :P |
| 03:14.52 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 03:15.09 | [TK]D-Fender | MrTelephone, I presume so. |
| 03:18.15 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 03:19.17 | *** join/#asterisk SirThomas_Home (n=tomc@209-169-199-174.us.warpdriveonline.com) |
| 03:22.13 | ZaVoid | meh everyversion i have crashes |
| 03:22.17 | ZaVoid | under load |
| 03:24.56 | sponger | exit |
| 03:27.12 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 03:29.11 | MrTelephone | the polycom phones don't tag the PC taffic with a vlanid |
| 03:33.19 | [TK]D-Fender | MrTelephone, meaning the PC is left to deal with its OWN LVAN settings |
| 03:33.56 | MrTelephone | it said the pc will be on the native vlan which is what is configured on the switch |
| 03:34.20 | MrTelephone | but if you configure port 18 for vlan 3 and u set the phone for vlan 4 traffic.. would the switch overwrite the vlan or disgard the frame? |
| 03:35.01 | *** join/#asterisk Cabal_ (n=Cabal@unaffiliated/cabal) |
| 03:35.19 | MrTelephone | from the frame |
| 03:35.35 | MrTelephone | i have some linksys managed switch so im downloading the manual to find out |
| 03:36.02 | MrTelephone | I had a problem with vlans earlier where the cable modem was seeing macs on the other side of a swithc with a different vlan |
| 03:36.39 | JT | heh, only a lowend managed switch i guess |
| 03:37.14 | MrTelephone | well you still think it would work somewhat |
| 03:37.30 | *** join/#asterisk elGuille (n=guillerm@200.69.237.107) |
| 03:37.43 | elGuille | elo |
| 03:37.46 | elGuille | helo, sorry |
| 03:37.56 | MrTelephone | no making fun jt |
| 03:38.02 | JT | hello if you want to be technical :) |
| 03:38.06 | MrTelephone | its a poe switch that was 700 bucks |
| 03:38.14 | JT | bargain ;) |
| 03:38.17 | MrTelephone | ciscos version was 3 grand |
| 03:38.19 | MrTelephone | haha |
| 03:38.25 | JT | everything that does PoE costs heaps at the moment |
| 03:38.39 | MrTelephone | 24 port poe is 700 |
| 03:38.58 | MrTelephone | i went to a cisco thing and they said the new wireless protocol eats up more juice than poe will allow |
| 03:39.07 | JT | i like procurve but damn it's expensive |
| 03:39.13 | MrTelephone | never heard of it |
| 03:39.19 | JT | hp procurve |
| 03:39.29 | MrTelephone | oh yeah I like those too |
| 03:39.30 | JT | the only switches i know of with lifetime warranties |
| 03:39.33 | MrTelephone | hp crap lasts forever |
| 03:41.02 | MrTelephone | the cisco guy had a webcam feed coming through his 7960 |
| 03:41.11 | MrTelephone | is that just xml to tell the phone to goto a certain webpage? |
| 03:41.34 | JT | either that or it's a video phone |
| 03:41.40 | JT | i think the 7960 is video |
| 03:42.10 | *** join/#asterisk bbryant_ (n=Brett@user-24-214-124-177.knology.net) |
| 03:42.37 | MrTelephone | the cisco guy said they programed it in "XML" |
| 03:42.41 | JT | sure |
| 03:42.45 | MrTelephone | i thought XML was just a bunch of settings |
| 03:42.56 | JT | it's a markup language |
| 03:43.01 | JT | some people using it for settings |
| 03:43.25 | MrTelephone | but you can only do what the phone will support, as for settings? |
| 03:43.31 | MrTelephone | unless you make your own firmware too? |
| 03:43.49 | [TK]D-Fender | 24 Port PoE = $370 USD <- |
| 03:44.00 | JT | obviously you can only do something that the phone is capable of |
| 03:44.17 | MrTelephone | true enough |
| 03:44.25 | MrTelephone | it will only do what it can do@ |
| 03:44.29 | MrTelephone | haha |
| 03:44.45 | *** join/#asterisk Rakko (n=eric@71-82-214-160.dhcp.mdsn.wi.charter.com) |
| 03:46.09 | MrTelephone | linksys manual download has been broken for days its really starting to piss me off |
| 03:46.15 | MrTelephone | nevemrind it load |
| 03:46.17 | MrTelephone | ed |
| 03:49.05 | Rakko | hi |
| 03:52.52 | *** join/#asterisk Bilbolodz (n=bilbo@pc-bilbo.man.lodz.pl) |
| 03:53.30 | Bilbolodz | Hi all |
| 03:53.58 | *** join/#asterisk bmg505 (n=leon@196.209.182.200) |
| 03:54.13 | Bilbolodz | I need help with asterisk 1.4 and codec transcoding |
| 03:58.33 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 03:58.54 | _VoiceMeUp_COM | heu |
| 03:58.58 | _VoiceMeUp_COM | 7960 not video mate |
| 03:59.01 | _VoiceMeUp_COM | 7970 maybe |
| 03:59.32 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:03.41 | Bilbolodz | My problem is: Asterisk 1.4.5 when codes conversion is needed (G726-G711a/u) sound is choppy. Asterisk 1.2.19 on the same server with near the same configuration is working properlly |
| 04:05.58 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:06.29 | Bilbolodz | More detail in my case: http://forums.digium.com/viewtopic.php?t=16415 |
| 04:06.34 | Bilbolodz | Please help me |
| 04:07.03 | ^Rocket^ | [TK]D-Fender so far so, got the zyxel to register and call VM, thanks for your help |
| 04:07.14 | ^Rocket^ | [TK]D-Fender I'll reread chap 5 tomorrow |
| 04:10.15 | *** join/#asterisk jetlagmk2 (n=jetlag@pool-70-106-87-151.hag.east.verizon.net) |
| 04:27.27 | *** join/#asterisk kn0x (n=pinochle@76.76.10.159) |
| 04:27.52 | kn0x | anyone have know how to get ztdummy working in a openVZ VPS? |
| 04:28.10 | JT | pray? |
| 04:30.46 | kn0x | pfft. |
| 04:30.50 | kn0x | that doesnt help |
| 04:30.56 | kn0x | is it possible? |
| 04:31.31 | JT | virtualisation often doesn't play nice with ast |
| 04:31.44 | kn0x | see i tried to switch to openpbx/callweaver, but then I couldn't use g729 :( |
| 04:32.00 | kn0x | so i have to choose between g729 and meetme |
| 04:32.40 | *** join/#asterisk hi365_m (i=HydraIRC@212.199.22.234.forward.012.net.il) |
| 04:32.44 | JT | choose between them? |
| 04:33.05 | hi365_m | hello. does asterisk 1.4 write queue log's to mysql? |
| 04:33.28 | kn0x | JT- yes |
| 04:33.35 | kn0x | I need meetme AND g729 |
| 04:33.58 | kn0x | openpbx's Nconference was a perfect meetme alternative, but I cant do g729 on opbx |
| 04:34.40 | kn0x | and I can't get ztdummy to work for me so I cant use asterisk w/ meetme |
| 04:35.14 | JT | must it run in openvz? |
| 04:35.37 | kn0x | yes, my thats my VPS |
| 04:36.10 | JT | well you have a few options |
| 04:36.25 | JT | either offload the conferencing (or g.729) elsewhere |
| 04:36.29 | JT | use app_conference |
| 04:36.34 | JT | or get ztdummy to work |
| 04:40.02 | kn0x | app_conference doesn't do announcements,kick/mute,ec. |
| 04:40.04 | kn0x | *etc |
| 04:40.15 | kn0x | those r the features I need |
| 04:41.42 | kn0x | JT, does app_conference support those features? |
| 04:41.49 | *** join/#asterisk Rakko (n=eric@71-82-214-160.dhcp.mdsn.wi.charter.com) |
| 04:42.21 | hi365_m | does asterisk 1.4 write queue log's to mysql? |
| 04:44.26 | Strom_M | hahahahaah, this is so cool - I'm finally getting the appliance to do things :) |
| 04:45.37 | *** join/#asterisk sid (n=unstable@tor/regular/sid) |
| 04:46.04 | sid | I wanted someone from asterisk/involved with asterisk to come to my local GNU/Linux Users Group in New York and talk about asterisk. Where is a good place to ask? |
| 04:46.18 | sid | Should I just hit the mailing list, or call up digium.. or what do you guys recommend? |
| 04:46.39 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 04:47.41 | JT | sid: the best bet would be to find people who use asterisk in new york |
| 04:48.01 | sid | JT I didn't see that listed on http://www.asterisk.org/community |
| 04:49.21 | JT | see what listed? |
| 04:49.55 | sid | An asterisk New York user group |
| 04:50.23 | sid | o, you mean find people who user asterisk in New York.. that's what I'm doing now. |
| 04:52.04 | JT | right :) |
| 04:52.18 | JT | there aren't many asterisk user groups around anyway |
| 04:53.00 | Strom_M | I live in Los Angeles, but hell, if you fly me out, I'm happy to talk to you guys :) |
| 04:53.58 | *** part/#asterisk hi365_m (i=HydraIRC@212.199.22.234.forward.012.net.il) |
| 04:59.21 | *** join/#asterisk steliosk (n=Stelios@62.169.217.209) |
| 05:03.24 | *** join/#asterisk antlers (n=antlers@ip70-173-89-173.lv.lv.cox.net) |
| 05:04.12 | russellb | sid: /wi9 |
| 05:04.17 | russellb | d'oh |
| 05:04.24 | russellb | ignore that. |
| 05:06.03 | russellb | quick, i need a feature request, i feel like coding ... |
| 05:06.47 | JT | fax |
| 05:07.28 | _VoiceMeUp_COM | lol |
| 05:07.32 | _VoiceMeUp_COM | was a fast asnwer |
| 05:08.27 | russellb | how about something i can do in an hour or less :-p |
| 05:08.36 | russellb | or, at all ... |
| 05:08.48 | JT | heh |
| 05:09.15 | jql | yes, count me in for a fax vote |
| 05:09.59 | russellb | gah, you people and your serious requests ... |
| 05:10.14 | jql | You wanted something fun? |
| 05:10.16 | jql | I got that |
| 05:10.39 | jql | Make the hold music take a [list of] filename[s] |
| 05:10.51 | russellb | instead of a directory? |
| 05:10.54 | jql | yeah |
| 05:11.08 | russellb | what's wrong with creating a directory and putting symlinks in it? :) |
| 05:11.21 | jql | and, perhaps even a dynamic program |
| 05:11.22 | jql | wheeee |
| 05:11.34 | JT | maybe it could regonise .pls files? ;) |
| 05:11.41 | jql | sox -abcdef file.mystery | |
| 05:11.56 | _VoiceMeUp_COM | app_callback |
| 05:12.41 | jql | I'm mainly cursing the music classes |
| 05:12.47 | _VoiceMeUp_COM | app_callback(CALLERID_AUTHORIZED,[PIN ], CALLBACK CONTEXT) |
| 05:12.53 | jql | since I can't actually generate them beforehand |
| 05:13.21 | JT | it should probably also specify the delay before callback |
| 05:13.26 | _VoiceMeUp_COM | yeah |
| 05:13.29 | _VoiceMeUp_COM | and retries |
| 05:13.46 | _VoiceMeUp_COM | czuse the whole call file crap is not to good.. chanllocal is bad and also well system too fast |
| 05:13.51 | russellb | you can do that with a System() call that echoes a few lines to a call file |
| 05:14.17 | russellb | what's wrong with call files? :) |
| 05:14.25 | _VoiceMeUp_COM | yeah but while your cell for example still sounds the congestion ( not to get biled by telco ) well asterisk is already sedning the call and getting a busy and your VM |
| 05:14.30 | _VoiceMeUp_COM | too fast |
| 05:14.45 | russellb | put a Wait in the callback extension before the dial |
| 05:14.49 | jql | I like call files. They're almost as useful as manager api business |
| 05:14.49 | _VoiceMeUp_COM | app call back should WAIT for hte current channel to be closed..wait 1 sec then call out and bridge nativelya ro wahtever |
| 05:14.51 | russellb | for like 5 seconds or whatever |
| 05:14.55 | _VoiceMeUp_COM | but not using LOCAL as channel |
| 05:14.55 | _VoiceMeUp_COM | etc |
| 05:15.10 | _VoiceMeUp_COM | i did |
| 05:15.10 | _VoiceMeUp_COM | ;) |
| 05:15.14 | russellb | heh |
| 05:15.23 | russellb | you could also write the call file in the 'h' extension |
| 05:15.37 | russellb | save off the parameters to channel variables to be written from there |
| 05:15.43 | _VoiceMeUp_COM | wait 7 |
| 05:15.50 | jql | yeah, I'd be using the h extension |
| 05:15.53 | russellb | that would wait until the call is down before doing the callback |
| 05:15.53 | _VoiceMeUp_COM | also weird stuff happens on chan local i think |
| 05:15.54 | _VoiceMeUp_COM | dtmf |
| 05:16.13 | _VoiceMeUp_COM | true |
| 05:16.38 | JT | one of the easiest ways to do reliable callback is with system() and a bash script |
| 05:16.50 | _VoiceMeUp_COM | waht i did |
| 05:17.00 | _VoiceMeUp_COM | DID,1,AGI(callbackagi.php) |
| 05:17.05 | _VoiceMeUp_COM | callback makes the file etc |
| 05:17.18 | _VoiceMeUp_COM | outs to callback ocntext.. waits 7 dials SER |
| 05:17.30 | jql | have the agi set a variable named CALLBACK_NUM |
| 05:17.36 | _VoiceMeUp_COM | other leg goes to tiemout digit 7 , response 10 , DISA |
| 05:17.47 | FreezeS | hey guys |
| 05:17.59 | FreezeS | I've got a big problem with queues |
| 05:18.11 | FreezeS | it adds a line to the CDR for every attempt to dial an agent |
| 05:18.26 | FreezeS | and I only need one line per incomming call |
| 05:18.28 | jql | FreezeS: Does the CDR have a call duration? |
| 05:18.42 | jql | because you need to ignore CDRs with a 0 duration regardless |
| 05:18.45 | russellb | i mean, they *are* attempted calls. |
| 05:18.49 | FreezeS | jql: sometimes it has 3 seconds |
| 05:18.55 | jql | okay, that bites |
| 05:18.58 | _VoiceMeUp_COM | ah i found why dtmf no go |
| 05:19.13 | FreezeS | but most of them are with NO ANSWER or BUSY |
| 05:19.16 | _VoiceMeUp_COM | using sip and instead of context/exten im using app and data ..with a macro |
| 05:19.26 | FreezeS | and it really messes up the CDRs |
| 05:20.04 | FreezeS | the line it adds for the incomming call is perfect, I get the CLI, the queue and the answering agent |
| 05:20.16 | _VoiceMeUp_COM | make the call NOCDR or whatever or amaflag(documentation) and act on that |
| 05:20.17 | FreezeS | I just don't need all that garbage in the cdrs |
| 05:20.38 | _VoiceMeUp_COM | show application NoCDR |
| 05:20.43 | _VoiceMeUp_COM | right before the Queue( |
| 05:20.54 | FreezeS | _VoiceMeUp_COM: I had about 3 NoCDR calls in the dialed context |
| 05:21.00 | FreezeS | and I still got CDRs for it :( |
| 05:21.05 | _VoiceMeUp_COM | wow |
| 05:21.07 | _VoiceMeUp_COM | lol |
| 05:21.13 | _VoiceMeUp_COM | some one should say the inverse would be bad |
| 05:21.16 | _VoiceMeUp_COM | but in your case |
| 05:21.19 | FreezeS | then I changed it to dial only sip |
| 05:21.32 | FreezeS | and still getting the same amount of CDRs |
| 05:21.41 | _VoiceMeUp_COM | ah you using lcoal in queues ? |
| 05:21.45 | _VoiceMeUp_COM | local i mean |
| 05:21.49 | FreezeS | first I did that |
| 05:22.00 | FreezeS | now I'm using sip\user |
| 05:22.08 | FreezeS | thought it will fix it |
| 05:22.19 | FreezeS | although I still need local |
| 05:22.29 | FreezeS | I have a setup with 3 boxes |
| 05:22.57 | _VoiceMeUp_COM | you have no C in the dial options to get there right ? |
| 05:23.27 | FreezeS | what do you mean ? |
| 05:23.47 | _VoiceMeUp_COM | nevermind lol waht your AST verison |
| 05:23.53 | FreezeS | 1.4.5 |
| 05:24.05 | FreezeS | bleeding edge technology :) |
| 05:25.04 | _VoiceMeUp_COM | weird |
| 05:25.09 | jql | The bleeding edge is the awesome edge |
| 05:25.55 | FreezeS | and another problem is that on theese CDRs I don't even have a cli |
| 05:26.10 | FreezeS | and I don't know to what queue they actually belong to |
| 05:26.18 | FreezeS | the cli was there in 1.4.0 |
| 05:26.30 | FreezeS | but when I upgraded, they thought it isn't necesarry |
| 05:35.45 | _VoiceMeUp_COM | a normal ping is 1 sec interval right ? |
| 05:36.59 | _VoiceMeUp_COM | hehe reboot time |
| 05:37.24 | _VoiceMeUp_COM | man some of these boxes up for 60 days.. i guess leakage could be cleaned .. 48 sec reboots is good .. |
| 05:37.51 | [hC] | i should probably reboot some of my main soft switches |
| 05:37.55 | [hC] | theyve been up for ~300 days |
| 05:37.55 | _VoiceMeUp_COM | yeah |
| 05:37.58 | _VoiceMeUp_COM | lol |
| 05:38.00 | [hC] | i wonder how im doing with memory leaks. |
| 05:38.02 | _VoiceMeUp_COM | talk about leaks |
| 05:38.09 | _VoiceMeUp_COM | AST 1.2.17 is not leaking at leas |
| 05:38.17 | [hC] | it went from i think.. 1.2.4 up to 1.2.15 |
| 05:40.28 | _VoiceMeUp_COM | VIRT 16688 RES 7088 SHR 4236 |
| 05:41.24 | _VoiceMeUp_COM | openpbx 109652 6876 |
| 05:41.27 | _VoiceMeUp_COM | lots more |
| 05:41.50 | _VoiceMeUp_COM | 25 process of SER's 51472 16096 |
| 05:41.52 | _VoiceMeUp_COM | hmm |
| 05:42.00 | JT | [hC]: they run asterisk? |
| 05:42.55 | [hC] | JT: what? |
| 05:43.08 | JT | the softswitches? |
| 05:43.27 | [hC] | JT: the boxes i use to act as a voice gateway to and from all my clients and the pstn (via pri or sip/iax) yes, run asterisk |
| 05:43.39 | [hC] | as opposed to a cisco or something. |
| 05:43.48 | _VoiceMeUp_COM | and VM and IVR |
| 05:43.50 | _VoiceMeUp_COM | etc |
| 05:43.54 | [hC] | well no |
| 05:43.57 | JT | surprised they stay up for 300days |
| 05:44.00 | [hC] | my clients all have their own pbx |
| 05:44.02 | _VoiceMeUp_COM | we use cisco /yate/freeswitch/opb and asterisk |
| 05:44.02 | JT | with no asterisk restarts |
| 05:44.08 | _VoiceMeUp_COM | i guess if another one comes out ill find a use for i |
| 05:44.10 | _VoiceMeUp_COM | it |
| 05:44.31 | [hC] | the box im talking about runs asterisk and all it does is handoffs to PRI or clients via IAX |
| 05:44.38 | [hC] | or, voip peers for some LD termination via SIP |
| 05:44.50 | [hC] | and nope, no need for asterisk restarts yet. |
| 05:45.04 | JT | lucky :) |
| 05:45.05 | [hC] | the box does nothing but act as a router for all calls. |
| 05:45.05 | _VoiceMeUp_COM | yeah |
| 05:45.12 | _VoiceMeUp_COM | think zaptel needs a refresh once in a while |
| 05:45.15 | [hC] | actually it does some light ivr too |
| 05:45.17 | _VoiceMeUp_COM | or wanrouter |
| 05:45.23 | [hC] | i use sangoma |
| 05:45.26 | [hC] | havent had to restart it |
| 05:45.32 | _VoiceMeUp_COM | yeah me neither stable |
| 05:45.32 | [hC] | ive upgraded the driver a couple times |
| 05:45.34 | _VoiceMeUp_COM | what version ? |
| 05:45.41 | [hC] | lets see what am i running right now |
| 05:45.42 | [hC] | sec. |
| 05:45.46 | _VoiceMeUp_COM | WANPIPE Release: 2.3.4-4 |
| 05:46.02 | [hC] | <PROTECTED> |
| 05:46.08 | _VoiceMeUp_COM | need to upgrade its -10 now |
| 05:46.10 | _VoiceMeUp_COM | but im scared |
| 05:46.12 | [hC] | WANPIPE Release: 2.3.4-4 |
| 05:46.18 | _VoiceMeUp_COM | hate to recompile all if all fails |
| 05:46.23 | [hC] | I average 20-25 calls a day via PRI |
| 05:46.28 | _VoiceMeUp_COM | yeah 4-4 isstable for me so no need to upg |
| 05:46.31 | [hC] | sorry |
| 05:46.33 | [hC] | not a day |
| 05:46.33 | _VoiceMeUp_COM | hee |
| 05:46.34 | [hC] | concurrent |
| 05:46.39 | JT | oh, that's very low volume |
| 05:46.40 | JT | ah |
| 05:46.40 | _VoiceMeUp_COM | was like WTh |
| 05:46.41 | [hC] | per day |
| 05:46.50 | [hC] | I usually fill a pri and a half with this box at peak |
| 05:46.55 | [hC] | average channel usage is 10-20 channel range. |
| 05:47.01 | [hC] | thats pri only |
| 05:47.04 | _VoiceMeUp_COM | waht area you in ? |
| 05:47.09 | [hC] | canada. vancouver. |
| 05:47.16 | _VoiceMeUp_COM | cool |
| 05:47.24 | _VoiceMeUp_COM | wished you where in sherbrooke QC |
| 05:47.31 | _VoiceMeUp_COM | that darn 819 is our next stop |
| 05:47.31 | *** part/#asterisk sid (n=unstable@tor/regular/sid) |
| 05:47.37 | _VoiceMeUp_COM | no one and i mean no one has qual there |
| 05:47.45 | [hC] | looks like i restarted asterisk on this box a week ago |
| 05:47.51 | [hC] | oh thats when i added another pri |
| 05:47.56 | [hC] | wonder what it was before that.. heh! |
| 05:48.03 | JT | do you use NFAS? |
| 05:48.05 | [hC] | im running 1.2.9.1 |
| 05:48.15 | [hC] | JT: I guess not, I dont know what NFAS is. |
| 05:48.27 | [hC] | maybe i just dont know the acronym |
| 05:48.30 | JT | Non Facility Associated Signalling |
| 05:48.38 | JT | shared D channels amongst PRIs |
| 05:48.41 | [hC] | no i dont |
| 05:48.42 | [hC] | on purpose. |
| 05:48.57 | [hC] | i have 1 dchannel per pri, but the telco rolls over from one to the other for inbound |
| 05:49.01 | _VoiceMeUp_COM | its neat |
| 05:49.02 | _VoiceMeUp_COM | nfas |
| 05:49.09 | _VoiceMeUp_COM | http://www.cisco.com/univercd/cc/td/doc/product/software/ios113ed/113t/113t_3/nfas.htm |
| 05:49.09 | [hC] | so in an emergency i can move the second pri to another box |
| 05:49.24 | _VoiceMeUp_COM | its like ss7 live call switching to another free B channel in case one pri goes down |
| 05:49.34 | _VoiceMeUp_COM | Any hard failure causes a switchover to the backup D channel and currently connected calls remain connected. |
| 05:49.34 | JT | err |
| 05:49.45 | _VoiceMeUp_COM | Use of a single D channel to control multiple PRI interfaces can free one B channel on each interface to carry other traffic. |
| 05:49.47 | [hC] | i didnt wawnt to have a restriction where both pri's relied ont he same dchannel and were not separable |
| 05:49.51 | *** join/#asterisk kaycut (n=nada@host171.190-31-153.telecom.net.ar) |
| 05:49.57 | kaycut | hi |
| 05:49.59 | _VoiceMeUp_COM | dont think they are |
| 05:49.59 | kaycut | everyone |
| 05:50.02 | JT | are you sure it can live switch an active call if a pri fails? |
| 05:50.06 | _VoiceMeUp_COM | thin kthey grou pthe D's in a gorup |
| 05:50.08 | *** join/#asterisk unfo (n=j@CPE000d8824ef4e-CM0013718690da.cpe.net.cable.rogers.com) |
| 05:50.11 | _VoiceMeUp_COM | and they can talk to eachotehr |
| 05:50.13 | [hC] | maybe i just dont understand it well enough, but it sounded scary. |
| 05:50.18 | kaycut | i need an answer |
| 05:50.23 | JT | you need a minimum of 2 D channels for NFAS |
| 05:50.27 | _VoiceMeUp_COM | JT check my link |
| 05:50.30 | [hC] | the telco had asked me if i wanted to have both pri's "share a dchannel" or each have their own |
| 05:50.35 | [hC] | i opted for independent |
| 05:50.45 | JT | but i see the advantage of [hC]'s setup |
| 05:50.54 | JT | especially if the telco fixes stuff at their end |
| 05:51.01 | _VoiceMeUp_COM | in an NFAS group with a primary D channel and a backup D |
| 05:51.02 | [hC] | yep |
| 05:51.08 | kaycut | in asterisk when a call is stablish it use server bandwith? |
| 05:51.15 | JT | _VoiceMeUp_COM: i doubt asterisk supports live B switching |
| 05:51.19 | _VoiceMeUp_COM | so i guess PRI 1 with D and uses pri 2 channel D as backup |
| 05:51.20 | [hC] | i dont think i have channel failover on their end, i just have rollover to the second pri's first bchannel if the first pri fills up. |
| 05:51.23 | _VoiceMeUp_COM | and reverse for PRI2 etc |
| 05:51.37 | _VoiceMeUp_COM | well if its hardware |
| 05:51.40 | _VoiceMeUp_COM | maybe no ? |
| 05:51.42 | JT | _VoiceMeUp_COM: no, one is primary for the group, the others are backups |
| 05:51.50 | _VoiceMeUp_COM | like on the zaptel layer or wanpipe ? |
| 05:51.51 | [hC] | I'm a network engineer turned voice guy, so im learning some of this stuff as i go :) |
| 05:52.06 | kaycut | anyone? |
| 05:52.07 | _VoiceMeUp_COM | ah its like instaead of 10 pris |
| 05:52.08 | JT | you can only start getting more B channels on traffic per PRI at > 2 PRIs |
| 05:52.12 | _VoiceMeUp_COM | you put 1 big with 230 channels |
| 05:52.18 | _VoiceMeUp_COM | and 10 d's all backed to eachother |
| 05:52.26 | JT | s/on/of/ |
| 05:52.29 | _VoiceMeUp_COM | so if anyone goes down thye reassign to liv eones |
| 05:52.48 | JT | yeah but people often get extra B channels in large groups, instead of unnecessary Ds |
| 05:52.49 | _VoiceMeUp_COM | ok |
| 05:53.00 | _VoiceMeUp_COM | ah true |
| 05:53.06 | _VoiceMeUp_COM | since its the D that ocntrolls all |
| 05:53.57 | *** join/#asterisk LooOD (n=gman@mamz.colo247.com) |
| 05:54.05 | [hC] | i need to pick up a book or find a resource online to learn more about the technology behind telco circuits |
| 05:54.09 | _VoiceMeUp_COM | well |
| 05:54.09 | kaycut | is there any form to do not use server bandwith in a call? |
| 05:54.18 | [hC] | i understand IP, ethernet, fiber, and routing protocols. :) |
| 05:54.19 | _VoiceMeUp_COM | i think its better to have the caririer hav ehtem failover |
| 05:54.32 | _VoiceMeUp_COM | so.. if pri1 goes down.. they send traffic to other btn |
| 05:54.37 | [hC] | kaycut: i have no idea what you're asking. |
| 05:54.47 | kaycut | ok |
| 05:54.55 | kaycut | i will came back tomorrow |
| 05:54.59 | kaycut | see yous |
| 05:55.14 | [hC] | ... i just meant i didnt understand his english, but okay |
| 05:55.37 | LooOD | Using xlite, when I check voice, it just calls myself and ask me to leave a message. Any ideas what I configured wrong? |
| 05:55.55 | LooOD | voice=voicemail |
| 05:56.32 | JT | [hC]: i suppose the painful way would be to go to itu.int and start reading recommendations (standards) cover to cover ;) |
| 05:56.48 | [hC] | ya.... no thanks :) |
| 05:56.50 | JT | there's a few good books available |
| 05:58.36 | *** join/#asterisk breanna_ (n=brea@c-76-23-9-101.hsd1.ut.comcast.net) |
| 05:58.55 | JT | probably depends what you want to learn too |
| 05:59.17 | breanna_ | What cool things will be in 1.6? |
| 05:59.55 | *** join/#asterisk RyanW (n=cableguy@ge0-0-15-lns0.207alg.qx21.net) |
| 06:00.44 | RyanW | Hello, i have a few snom 360's that i've setup using http provisioning, when the phones boot up they prompt the users to enter a password. If the user hits OK it will alter the sip password and cause the phone to not register. |
| 06:01.15 | RyanW | How do i stop my phones prompting the user and just use the sip password as specified in the http provisioning. |
| 06:02.33 | ltdwk | if it is prompting them it's because they didnt correctly receive their line settings |
| 06:03.06 | ltdwk | but you should also set |
| 06:03.10 | ltdwk | logon_wizard: off |
| 06:03.16 | ltdwk | in your provisinoing config |
| 06:03.41 | RyanW | thanks, mine was set to on. |
| 06:05.34 | RyanW | http://pastebin.ca/580700 is the provisioning, i disabled the logon wizard but they're still prompting |
| 06:05.51 | `Sean | Hey |
| 06:06.04 | `Sean | whats was that Area code not listed in NANPA the goverment One |
| 06:06.06 | `Sean | i keep frogeting |
| 06:06.07 | ltdwk | like i said |
| 06:06.16 | ltdwk | your info isnt getting to the phones |
| 06:06.19 | *** part/#asterisk unfo (n=j@CPE000d8824ef4e-CM0013718690da.cpe.net.cable.rogers.com) |
| 06:06.48 | RyanW | ltdwk, i just updated the logon wizard setting through my provisioning and saw it take effect with a refresh of the settings.htm page in my phone |
| 06:06.48 | ltdwk | you shouldn't have the & inside your config |
| 06:06.51 | ltdwk | or $ |
| 06:07.01 | _VoiceMeUp_COM | Sean you found a place to hack with forged accounts ? |
| 06:07.07 | _VoiceMeUp_COM | ;) |
| 06:07.13 | `Sean | HUh? |
| 06:07.18 | _VoiceMeUp_COM | j/k |
| 06:07.31 | `Sean | IM trying to get the US goverment Area code |
| 06:07.33 | `Sean | i forgot what it was |
| 06:07.33 | _VoiceMeUp_COM | soudned like a hacker leet question |
| 06:08.01 | RyanW | ltdwk, i just removed the & from my config but its still prompting. |
| 06:08.08 | Strom_M | `Sean: 700 |
| 06:08.12 | Strom_M | er, no |
| 06:08.14 | Strom_M | 710 :) |
| 06:08.20 | _VoiceMeUp_COM | pm me |
| 06:08.22 | _VoiceMeUp_COM | sean |
| 06:08.22 | ltdwk | ryanw: the & in the "settings" page indicates the setting was obtained from provisioning, do you see them in your config for the setings in that file ? |
| 06:08.47 | `Sean | thanks steliosk |
| 06:08.50 | `Sean | etr |
| 06:08.52 | `Sean | thanks Strom_M |
| 06:09.47 | *** join/#asterisk waptaxi (n=waptaxi@45.151-224-87.telenet.ru) |
| 06:10.17 | `Sean | Area code 710 is the United States area code reserved to the Federal government of the United States. As of December 2006 it has only one working number, 710-NCS-GETS (710-627-4387), which requires a special access code to use. |
| 06:10.45 | RyanW | i read on the snom wiki that ! means reset at boot, & means wont reset at boot and there was another symbol which meant read only. |
| 06:11.35 | ltdwk | ryanw: my experience leads me to believe that & indicates the setting was obtained from provisioning |
| 06:12.27 | RyanW | ltdwk, I'll go investigate and get back to you, i think if you don't specify either a $ or & in your config file then & will be the default |
| 06:12.57 | ltdwk | ryanw: possibly |
| 06:13.16 | ltdwk | ryanw: settings obtained from provisioning are read only by default I think |
| 06:14.13 | ltdwk | ryanw: either way it's a good way to tell if your settings have actually come from the provisioning config |
| 06:14.26 | *** join/#asterisk Aces1Up (n=really@ip68-227-41-148.lv.lv.cox.net) |
| 06:14.51 | ltdwk | i found the only times i ever had the logon wizard pop up were when the phone was not able to obtain its settings from the server |
| 06:15.00 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 06:15.11 | ltdwk | (specifically the user line settings) |
| 06:15.30 | ltdwk | as soon as they are set, the logon wizard never pops up |
| 06:17.39 | RyanW | ltdwk, i just discovered that one my phones is now working but some others aren't and your suspission was correct, its not updating from the server. |
| 06:18.11 | ltdwk | check to make sure you've programmed the MACs correctly into your scripts or file names |
| 06:18.29 | RyanW | ltdwk, http://snom.com/wiki/index.php/Functions/Phone/Mass_deployment "Hints for setting files" covers the various symbols |
| 06:19.06 | ltdwk | ryanw: yeah... i've read that many times... however, in my experience the only things i set in provisioning are the ones i don't want changed |
| 06:19.48 | ltdwk | and there's no way through the gui to make a setting read only i've seen, so all settings, unless otherwise specified with ! or $, will be made read only from provisioning |
| 06:20.20 | RyanW | my users are going to be less tech-savy so i'm going to fix some settings and set some others to default but allow the users to change them. |
| 06:25.54 | ltdwk | mine are in no way tech savvy which is why i make everything read only |
| 06:27.37 | LooOD | How are you suppose to check your voicemail? |
| 06:30.12 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 06:36.49 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 06:43.44 | Bilbolodz | My problem is: Asterisk 1.4.5 when codes conversion is needed (G726-G711a/u) sound is choppy. Asterisk 1.2.19 on the same server with near the same configuration is working properlly, can anynone help me? |
| 06:46.44 | *** part/#asterisk philippel (n=p_lindhe@c-24-17-254-189.hsd1.mn.comcast.net) |
| 06:53.35 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:59.28 | DarKnesS_WolF | i have something in mind .. now let us say that i have one sip phone with one sip accountcode and i want to assgin many passwords for the same phone.... |
| 06:59.31 | DarKnesS_WolF | how to do so ? |
| 06:59.43 | DarKnesS_WolF | can the sip account has many account codes ? |
| 07:00.27 | Strom_M | why would you want to assign many passwords? |
| 07:01.01 | *** join/#asterisk yassaccan (n=yassacca@admin175.hgo.se) |
| 07:01.33 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 07:01.49 | DarKnesS_WolF | so many users will use it and then i can track back there calls |
| 07:02.00 | DarKnesS_WolF | the idea is how i can check this into the CDR... |
| 07:02.18 | Strom_M | um |
| 07:02.25 | Strom_M | you want to give each user their own account |
| 07:02.42 | DarKnesS_WolF | what do u mean ? |
| 07:02.52 | DarKnesS_WolF | i have only one SIP phone with one sip account they will use... |
| 07:02.59 | _VoiceMeUp_COM | ok |
| 07:03.05 | _VoiceMeUp_COM | user CDRUSERFIELD |
| 07:03.08 | _VoiceMeUp_COM | i mean user |
| 07:03.10 | Strom_M | and they're all using the same physical telephone set? |
| 07:03.10 | _VoiceMeUp_COM | i mean use |
| 07:03.13 | *** join/#asterisk Polis_ttt (n=your@194-237-172-225-no48.business.telia.com) |
| 07:03.17 | DarKnesS_WolF | Strom_M: yes |
| 07:03.28 | _VoiceMeUp_COM | ah no.. hmm |
| 07:03.29 | DarKnesS_WolF | _VoiceMeUp_COM: yes i have this idea but i want to check if there is anyother way .. |
| 07:03.33 | Strom_M | is this phone in a brothel or something? |
| 07:03.56 | DarKnesS_WolF | Strom_M: in what ? |
| 07:03.59 | _VoiceMeUp_COM | wahts a brothel ? |
| 07:04.13 | _VoiceMeUp_COM | ah |
| 07:04.15 | _VoiceMeUp_COM | un bordel |
| 07:04.19 | _VoiceMeUp_COM | hoe house ? |
| 07:04.25 | Strom_M | yes |
| 07:04.29 | _VoiceMeUp_COM | http://en.wikipedia.org/wiki/Brothel |
| 07:04.33 | _VoiceMeUp_COM | first time i hear htat name |
| 07:04.50 | Strom_M | _VoiceMeUp_COM: where are you? |
| 07:05.00 | _VoiceMeUp_COM | The word brothel is from Middle English, and stems from from 'brothen', the past participle of 'brethen', meaning 'to waste away' or 'to go to ruin', |
| 07:05.03 | _VoiceMeUp_COM | lol Canada |
| 07:05.18 | _VoiceMeUp_COM | i guess that also defines Mariage |
| 07:05.37 | Strom_M | you know, I should buy you a dictionary or something |
| 07:05.51 | Strom_M | you don't know the word brothel and you can't spell marriage :) |
| 07:05.54 | _VoiceMeUp_COM | Thanks ;) its not my first language |
| 07:05.59 | Strom_M | ah ok |
| 07:06.05 | _VoiceMeUp_COM | yeah i spelled it in french |
| 07:06.32 | _VoiceMeUp_COM | the point of communication is to get yourself understood.. ;) |
| 07:06.46 | _VoiceMeUp_COM | tehe point of being a teacher is doing it well |
| 07:06.47 | _VoiceMeUp_COM | ;) |
| 07:07.02 | _VoiceMeUp_COM | i could actualy type i cdnuolt blveiee taht I cluod aulaclty uesdnatnrd waht I was rdanieg |
| 07:07.06 | _VoiceMeUp_COM | and you would understand |
| 07:07.12 | JT | WHAT |
| 07:07.15 | JT | :P |
| 07:07.52 | *** part/#asterisk harlequin516 (n=sham@styk.net) |
| 07:08.26 | Strom_M | uh yeah, that parses as gibberish |
| 07:10.17 | *** join/#asterisk achu (n=achu@122.167.63.163) |
| 07:13.43 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 07:18.16 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 07:18.19 | zeeesh | hi |
| 07:20.03 | *** join/#asterisk tengulre (n=tengulre@222.90.66.10) |
| 07:20.20 | *** join/#asterisk shay|work (n=shay@unaffiliated/shay) |
| 07:20.56 | *** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl) |
| 07:23.18 | zeeesh | nobody is there ? |
| 07:24.17 | Strom_M | nopw |
| 07:24.19 | Strom_M | all dead here |
| 07:24.32 | *** join/#asterisk af_ (n=getsmart@81-174-44-131.dynamic.ngi.it) |
| 07:34.26 | achu | can anybody help me to configure the jivetel trunk ? |
| 07:35.05 | tengulre | achu: what's the jivetel? a hardware? |
| 07:35.25 | achu | sorry its same as a broadvoice connection |
| 07:35.45 | achu | I mean a telephone line |
| 07:36.18 | achu | www.jivetel.com |
| 07:37.36 | *** join/#asterisk dikdust (n=dikdust@gandalf.ipv6.adfacom.it) |
| 07:37.58 | tengulre | I think people are sleeping! |
| 07:39.30 | achu | Dovid told me that jivetel is very good than broadvoice |
| 07:39.39 | achu | so I want to try that |
| 07:40.20 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-67-180-78-55.hsd1.ca.comcast.net) |
| 07:41.02 | zeeesh | right now if u guys superior than me |
| 07:41.21 | zeeesh | then will anybody guide about parameter "s" |
| 07:41.52 | zeeesh | coz i hv a DID .. configured at asterisk-1.4.4 |
| 07:42.17 | achu | The "s" extension is used when there is no known called number in the context used |
| 07:42.23 | zeeesh | i use this way ... for call out .. |
| 07:42.37 | achu | http://www.voip-info.org/wiki/index.php?page=Asterisk+s+extension |
| 07:42.49 | zeeesh | exten => 55555,1,Answer |
| 07:43.12 | zeeesh | exten => _X.,2,WaitExten(15) |
| 07:43.20 | JT | err |
| 07:43.41 | JT | you always need a 1 priority unless some other context calls a priority specifically |
| 07:44.03 | zeeesh | exten => _x.,3,Dial(sip/${EXTEN}@ibasis) |
| 07:44.12 | zeeesh | working fine |
| 07:44.24 | zeeesh | how to replace these with "s" |
| 07:44.28 | JT | you must have forgotten to paste the priority 1 line then |
| 07:51.40 | dikdust | hi, someone has tried to put italian language in 1.2.x ? For me doesn't work |
| 07:54.48 | snuffy22 | hmm.. is it just me or is digium.com down? |
| 07:55.03 | snuffy22 | asterisk.org works.. |
| 07:55.43 | JT | www.digium.com is negative function |
| 07:56.07 | snuffy22 | just when i wanted to look at bug ids too :( |
| 07:56.09 | *** join/#asterisk af_ (n=getsmart@81-174-45-51.dynamic.ngi.it) |
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| 08:02.57 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:03.29 | *** part/#asterisk achu (n=achu@122.167.63.163) |
| 08:04.09 | *** join/#asterisk bbryant_ (n=Brett@user-24-214-124-177.knology.net) |
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| 08:18.21 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 08:18.21 | *** topic/#asterisk is Asterisk: The Open Source Telephony Application Platform -=- Asterisk 1.2.19 and 1.4.5 (June 15, 2007) *-addons 1.2.7 and 1.4.2 (June 18, 2007) -=- Join #asterisknow or #asterisk-gui for AsteriskNOW and Asterisk-GUI info -=- Join #asterisk-commits to monitor svn changes -=- Join #freepbx for freepbx/#trixbox for trixbox support. |
| 08:43.37 | key2 | hey |
| 08:43.53 | key2 | if I have a .ko, how can I tell linux to automatically modprobe it when booting |
| 08:43.53 | key2 | ? |
| 08:46.05 | tzafrir | key2, on which distro? |
| 08:46.37 | tzafrir | e.g.: on debian, add it to /etc/modules |
| 08:47.39 | key2 | tzafrir: thats what I did |
| 08:48.27 | key2 | tzafir: mmh centos ? |
| 08:48.50 | tzafrir | which module is it? |
| 08:49.08 | tzafrir | I'm not aware of an equivalent |
| 08:49.51 | key2 | qozap |
| 08:49.53 | key2 | for BRI |
| 08:49.57 | key2 | with bristuff |
| 08:50.11 | key2 | so everytime am forced to rmmod all the zaptel, modprobe it |
| 09:01.49 | *** join/#asterisk berktr (n=cn@teknopet.com) |
| 09:01.51 | berktr | hello |
| 09:02.02 | berktr | what happens if i set nat=yes and use that peer without nat |
| 09:12.41 | *** join/#asterisk n3glv (n=n3glv@c-71-60-125-243.hsd1.pa.comcast.net) |
| 09:12.54 | n3glv | anyone here know polycom 500's ? |
| 09:13.37 | s0ck | lo n3glv |
| 09:13.46 | n3glv | hi |
| 09:13.52 | n3glv | I got a poly 500 |
| 09:13.58 | s0ck | and i don't, mi scuzi |
| 09:14.04 | n3glv | I think I have a dialplan issue |
| 09:14.05 | s0ck | having fun with it? :D |
| 09:14.08 | *** join/#asterisk matsk (n=mk@194.68.102.173) |
| 09:14.11 | n3glv | can't dial * prefix on calls |
| 09:14.11 | *** join/#asterisk mattfletcher (n=matt@88-97-179-134.dsl.zen.co.uk) |
| 09:14.24 | s0ck | what you use * for |
| 09:14.31 | n3glv | so, if I want features that start with star |
| 09:15.03 | n3glv | or whatever, I can't do them |
| 09:15.03 | s0ck | ah |
| 09:15.03 | s0ck | *79 etc |
| 09:15.03 | n3glv | asterisk 1.4.4 |
| 09:15.03 | n3glv | right, *43 etc |
| 09:15.03 | s0ck | not a dtmf issue is it? |
| 09:15.06 | n3glv | no |
| 09:15.11 | n3glv | go's into to: |
| 09:15.18 | n3glv | like trying to dial sip |
| 09:16.15 | s0ck | gl |
| 09:16.24 | mattfletcher | Can anyone answer this: If I receive a call, and then use the transfer button on my SIP phone (an aastra 480i) to put the call through to another extension, is there any way of retrieving the first call's caller id within the dial plan of the second call? |
| 09:16.38 | s0ck | i just hooked a box up to a bri circuit but the number refuses to ring inbound |
| 09:16.50 | s0ck | bt reckon an isdn circuit wont ring unless a handset/pbx is attached |
| 09:17.03 | s0ck | it is attached... so wondering if i got something wrong here |
| 09:17.11 | s0ck | b410p is showing a green light too on the port :s |
| 09:17.56 | n3glv | k |
| 09:17.57 | n3glv | tnx |
| 09:18.01 | *** part/#asterisk n3glv (n=n3glv@c-71-60-125-243.hsd1.pa.comcast.net) |
| 09:24.52 | *** join/#asterisk oej (n=olle@apollo.webway.se) |
| 09:24.53 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:25.26 | berktr | mattfletcher, tell me if you can figure out how because i was looking for the same |
| 09:25.29 | JT | s0ck: is anything coming up in the cli upon an incomikng call? |
| 09:27.29 | mattfletcher | berktr: i bet a lot of people are. i want my systray cid client to use the true incoming number rather than the receptionist's number. if i don't find a way, i'm thinking of hacking it a bit with DB sets on one call and somehow working out that the next call is made immedaitely after. Or something. |
| 09:27.42 | s0ck | JT: i see nothing for any incoming call |
| 09:27.54 | s0ck | you dial the number from outside, it doesn't even attempt to ring, just goes to a dead tone |
| 09:28.10 | s0ck | i was convinced it was a bt fault but the tech was telling me otherwise... |
| 09:29.55 | JT | i'm betting you are using pile of bugs, i mean misdn :P |
| 09:30.02 | s0ck | hehe |
| 09:30.09 | s0ck | following the digium guide |
| 09:30.14 | s0ck | it does indeed say to use misdn |
| 09:30.18 | JT | yeah |
| 09:30.19 | s0ck | didn't realise there was an alternative |
| 09:30.26 | JT | not sure if the digium card works with bristuff or not |
| 09:30.28 | *** join/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com) |
| 09:30.34 | s0ck | thing is |
| 09:30.34 | JT | haven't heard of anyone trying |
| 09:30.34 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 09:30.40 | s0ck | when i dial out |
| 09:30.50 | s0ck | i get a bt woman voice telling me the call can't be made |
| 09:30.55 | s0ck | definitely is not an asterisk voice |
| 09:32.19 | mattfletcher | berktr: running with my idea of storing the number in the DB, if you used a different number to transfer calls to (than making normal calls) you could make it a bit more solid |
| 09:32.59 | JT | s0ck: you might be sending the wrong digits in the dial out case |
| 09:33.34 | s0ck | that's what i wondered tbh |
| 09:34.34 | s0ck | misdn/1/${EXTEN} is what i have though |
| 09:34.42 | berktr | well mattfletcher |
| 09:34.49 | berktr | i am now brainstorming on papers here |
| 09:34.50 | s0ck | i dont believe it's a feature line although i've sent 9 with the number too |
| 09:35.06 | snuffy22 | mm... anyone get a warning when compiling 1.4.5.. somethin about 'vm_delete not being explictly defined' |
| 09:35.46 | berktr | even if you use a db, how are you going to fetch the correct data? |
| 09:36.06 | berktr | by using timestamps right |
| 09:36.16 | berktr | there is a chance for it to fail somehow |
| 09:36.23 | mattfletcher | really not sure. you could store the time and see how much time has lagged, not gonna be perfect |
| 09:36.51 | s0ck | 'Please hang up and try again' |
| 09:36.59 | mattfletcher | is there perhaps a way of seeing how many calls are in progress to a particular extension? |
| 09:37.11 | mattfletcher | if > 1 then assume a transfer |
| 09:37.16 | berktr | what do you mean |
| 09:38.05 | mattfletcher | presumably if a call comes in and is put on hold then another call is made, there must be some way of asterisk knowing that extension xxx has 2 calls "on the go" |
| 09:38.42 | s0ck | JT: for TE mode, a normal rj45 cable to the isdn2 box? |
| 09:40.08 | berktr | maybe mattfletcher, i'll think |
| 09:40.15 | berktr | another interesting question |
| 09:40.36 | berktr | is it possible to set different music on hold music to different dialing numbers |
| 09:40.54 | berktr | lets say 5173034899 is calling and i want that user to listen to santana when i put him on hold |
| 09:41.07 | berktr | when another number calls, i want him to listen to classical .. |
| 09:41.09 | mattfletcher | http://archives.free.net.ph/message/20060302.201752.fb57c8d7.en.html asks a similar qn to mine. no answers but maybe some inspiration |
| 09:41.56 | mattfletcher | i think you can define which class of music is used within a Dial() command. does this not perpetuate to a hold made after the dial? |
| 09:42.23 | berktr | well, how am i going to set the moh music according to the caller id |
| 09:44.00 | mattfletcher | erm, how many different caller id's would it need to recognise. if only a handful, you could set up different dial commands using GotoIf statements on the caller id |
| 09:44.09 | berktr | hmm |
| 09:44.19 | berktr | like 3-4 different caller ids |
| 09:44.21 | berktr | not much |
| 09:44.23 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.148.139) |
| 09:46.27 | mattfletcher | yeah so (pseudocode) gotoif(callerid=123456789,5) gotoif(callerid=987654321,6) then 5 and 6 would call the same extension but with different moh defined |
| 09:47.39 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 09:47.40 | berktr | i see |
| 09:50.16 | mattfletcher | with my problem, i wonder if there is any difference between pressing transfer and hanging up. if so, i could delete the DB setting on hangup maybe. |
| 09:53.13 | s0ck | YEY |
| 09:53.17 | s0ck | got it to dial something |
| 09:54.10 | JT | s0ck: correct, normal cable |
| 09:54.15 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com) |
| 09:55.24 | s0ck | JT: misdn/1/${EXTEN} <--- ${EXTEN} being the number i pass to it from the handset/softphone yeh? |
| 09:55.47 | s0ck | if i force the trunk to use misdn/1/07778344344 (my mobile) it dials it |
| 09:59.21 | DarKnesS_WolF | so i can't assgin multy accountcode to a sing SIP account ? |
| 10:03.36 | s0ck | ok |
| 10:03.38 | s0ck | sussed it |
| 10:03.44 | s0ck | $OUTNUM$ ftw |
| 10:06.41 | *** join/#asterisk HarryR (n=Administ@host-83-146-53-46.bulldogdsl.com) |
| 10:15.32 | mattfletcher | anyone, is it possible to set a db record, dial a number, and then delete the db record when the call hangs up? |
| 10:16.09 | mattfletcher | or after the Dial() is picked up does the dialplan stop taking effect? |
| 10:17.49 | JT | s0ck: are you using asterisk? |
| 10:18.01 | JT | ${EXTEN} should've been fine |
| 10:19.09 | Strom_M | mattfletcher: look at the h extension |
| 10:19.14 | sergee | s0ck: are you from Kazahstan? :) |
| 10:21.11 | *** join/#asterisk dseeb_ (n=dcb@124.177.45.198) |
| 10:21.28 | dseeb_ | <PROTECTED> |
| 10:21.34 | JT | my guess is england |
| 10:29.59 | CBU[^_^]M`` | my asterisk wont install on my computer .... |
| 10:31.53 | dseeb_ | why not? |
| 10:35.09 | Strom_M | no; surely we can determine the cause, the solution, and the recipe for your mother's cheesecake from that explanation alone |
| 10:38.21 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 10:38.22 | berktr | lol |
| 10:39.39 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 10:40.05 | sergee | JT: there is a cell operator in Kazahstan with code 777 |
| 10:43.48 | *** join/#asterisk Fl1p (n=david@195.14.211.55) |
| 10:44.27 | Fl1p | hi, when using the spool service asterisk said scan_service: Unable to open permission denied, call file is created with root and asterisk running as root |
| 10:46.46 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:47.05 | Fl1p | no idea anyone ? |
| 10:47.22 | Strom_M | clearly the answer is to add more cheesecake |
| 10:48.35 | *** join/#asterisk friedrich| (n=friedric@e177254221.adsl.alicedsl.de) |
| 10:49.27 | Fl1p | Ive created a call file which should create a call a number by asterisk, after copy to /var/spool/asterisk/outgoing the CLI says pbx_spool.c:347 scan_service: Unable to open /var/spool/asterisk/outgoing/call01.call: Permission denied, deleting |
| 10:49.28 | Fl1p | Jun 21 12:46:17 WARNING[13678]: pbx_spool.c:389 scan_thread: Failed to scan service '/var/spool/asterisk/outgoing/call01.call' |
| 10:49.49 | *** part/#asterisk Cybertoy (n=cybertoy@dsl254-123-112.nyc1.dsl.speakeasy.net) |
| 10:50.08 | Fl1p | permission of outgoing directory is 700 the call files are 644 |
| 10:50.51 | *** join/#asterisk penguinFunk_ (n=penguin@unaffiliated/penguinfunk) |
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| 10:55.56 | HarryR | Is anybody using chan_gtalk or chan_jingle? |
| 10:56.53 | DarKnesS_WolF | HarryR: yes i do |
| 10:57.19 | DarKnesS_WolF | any idea what this is m - Interpret the given path as a file which contains a list of account codes and password hashes delimited with ':', listed one per line in the file. When one of the passwords is matched, the channel will have its account code set to the corresponding account code in the file. in Authenticate cmd ? i can't get the fomrat of the file correctly |
| 10:57.43 | *** join/#asterisk plantseeker (n=chatzill@83.167.161.28) |
| 10:58.31 | HarryR | DarKnesS_WolF: having any problems with audio not being recieved or sent? |
| 10:59.38 | DarKnesS_WolF | HarryR: no everything works |
| 11:00.52 | HarryR | ah, must just be my computer then |
| 11:02.31 | DarKnesS_WolF | HarryR: getting in errors in the asterisk side ? |
| 11:05.18 | Strom_M | plz wat..... |
| 11:05.22 | Strom_M | PLZ !!!!! |
| 11:05.40 | *** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
| 11:09.40 | DarKnesS_WolF | Strom_M: what ? |
| 11:10.34 | *** part/#asterisk Fl1p (n=david@195.14.211.55) |
| 11:10.52 | Strom_M | PLZ <--- |
| 11:11.49 | Strom_M | ---> PLZ |
| 11:12.10 | Strom_M | -> PLZ < |
| 11:13.23 | *** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com) |
| 11:17.50 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 11:19.02 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 11:19.03 | zeeesh | hi |
| 11:19.17 | yonahw-work | hey if ztcfg runs without error and shows all the channels on a pri yet asterisk shows no zap channels is the problem neccesarily in my extensions.conf? |
| 11:19.40 | Strom_M | yonahw-work: or you completely forgot to configure zapata.conf |
| 11:21.13 | yonahw-work | nope zapata.conf is configured, i had it configured for a different pri line and have changed nothing |
| 11:21.49 | Strom_M | well if you changed nothing, then asterisk should still know about the channels |
| 11:21.56 | Strom_M | pastebin zapata and zaptel |
| 11:22.00 | Strom_M | pastebin.ca |
| 11:22.03 | yonahw-work | well i changed pri's |
| 11:22.08 | yonahw-work | will do thanks |
| 11:22.37 | pj_ | And you haven't got any error when you start asterisk ? |
| 11:22.39 | yonahw-work | do i need to configure trunks in zapata.conf? |
| 11:22.54 | yonahw-work | i just realized there are no trunks configured there |
| 11:22.56 | pj_ | oh yeah you do |
| 11:26.03 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 11:26.11 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 11:37.12 | yonahw-work | ok i posted my zaptel and zapata to http://pastebin.ca/581129 |
| 11:37.18 | DarKnesS_WolF | anyone got authenticate with option M working correctly ? |
| 11:38.06 | Strom_M | yonahw-work: you dont need to define trunkgroups unless you've got NFAS ISDN PRI T1s |
| 11:38.14 | Strom_M | and you have a typo in zapata.conf |
| 11:38.38 | Strom_M | so take that trunkgroup stuff out and fix this line: |
| 11:38.38 | Strom_M | channel =>1-15,17=31 |
| 11:38.44 | Strom_M | should be 17-31 |
| 11:38.56 | yonahw-work | oh my bad thanks |
| 11:38.58 | yonahw-work | let me try that |
| 11:41.57 | *** join/#asterisk gardo (n=gardo@121.97.194.205) |
| 11:42.09 | JT | people often seem to play with trunkgroups unnecessarily |
| 11:42.18 | Strom_M | come on, man, it doesnt take four minutes to reload chan_zap.so :) |
| 11:42.38 | Strom_M | JT: could it be...that....they don't know what they're doing?!!!?? |
| 11:42.56 | yonahw-work | i actually did not originally have the trunk group setup i just changed that now |
| 11:43.04 | yonahw-work | you hit on the buttom |
| 11:43.06 | JT | and when they get advice from others that is bad, like pj_ :P |
| 11:43.18 | yonahw-work | actually that was what made me change it |
| 11:43.23 | yonahw-work | still does not work though |
| 11:43.34 | JT | does not work, not terribly descritive |
| 11:43.38 | JT | descriptive |
| 11:43.45 | Strom_M | HALP IT WORKS NOTTT |
| 11:43.50 | yonahw-work | lol |
| 11:44.03 | Strom_M | FART ON SEX BONER |
| 11:44.15 | yonahw-work | i still get the same error of no channel type registered for zap |
| 11:44.25 | JT | erm |
| 11:44.29 | JT | so it's not loading |
| 11:44.37 | JT | watch for the errors on asterisk startup |
| 11:44.38 | *** join/#asterisk af_ (n=getsmart@81-174-45-51.dynamic.ngi.it) |
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| 12:00.19 | yonahw-work | i have a bunch of modules which are not registering one of which is app_zapscan.so i suspect this may be a hint |
| 12:01.42 | yonahw-work | strom, jt: i appreciate your guys pointing me in the right direction there, I will do a little more homework here and see where i get |
| 12:03.19 | JT | zapscan, doesn't sound like something you need |
| 12:03.50 | *** join/#asterisk matsk (n=mk@194.68.102.173) |
| 12:05.02 | *** part/#asterisk javar (n=javar@69.79.134.24) |
| 12:05.10 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 12:05.53 | s0ck | JT: yes |
| 12:08.28 | *** join/#asterisk sergee (i=kvirc@195.94.224.197) |
| 12:10.29 | *** part/#asterisk plantseeker (n=chatzill@83.167.161.28) |
| 12:14.39 | *** join/#asterisk guillote_GNU (n=guillote@host155.200-117-35.telecom.net.ar) |
| 12:15.18 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 12:15.42 | *** join/#asterisk psi0n (n=psi@133.80-202-238.nextgentel.com) |
| 12:15.42 | yonahw-work | hmm wonder what else it could be |
| 12:16.27 | rob0 | Weasels have eaten your Zap channels. |
| 12:16.45 | yonahw-work | rob0: I kid you not the thought has crossed my mind |
| 12:17.29 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 12:17.30 | cpm | <PROTECTED> |
| 12:17.41 | cpm | the ice weasels come, |
| 12:17.42 | cpm | please tell me what to do! |
| 12:17.55 | [TK]D-Fender | cpm : PRAY |
| 12:18.07 | cpm | [TK]D-Fender, noted, thanks |
| 12:18.31 | [TK]D-Fender | cpm: You'll "ferret" out the problem eventually ;) |
| 12:18.39 | coppice | where does that thing about the ice weasels originally come from? |
| 12:19.13 | yonahw-work | "chan_zap.so did not register itself during load" might this be related? |
| 12:19.19 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 12:19.36 | cpm | Matt Groenig I think, pre-simpsons |
| 12:20.06 | coppice | sounds about right for him |
| 12:20.37 | cpm | Ahh, here it is! |
| 12:20.44 | [TK]D-Fender | coppice: For Debian's renaming of Firefox due to philosophical / licensing disagreements.... |
| 12:20.49 | cpm | Love is a snowmobile racing across the tundra and then suddenly it flips over, pinning you underneath. At night, the ice weasels come. |
| 12:21.06 | cpm | --Friedrich Nietzsche |
| 12:21.18 | cpm | <PROTECTED> |
| 12:23.12 | coppice | well, Nietzche, Groenig, its much the same |
| 12:23.37 | DarKnesS_WolF | [TK]D-Fender: any idea about the authnteciate command with m option ? |
| 12:24.10 | DarKnesS_WolF | any idea what this is m - Interpret the given path as a file which contains a list of account codes and password hashes delimited with ':', listed one per line in the file. When one of the passwords is matched, the channel will have its account code set to the corresponding account code in the file. in Authenticate cmd ? i can't get the fomrat of the file correctly |
| 12:24.13 | [TK]D-Fender | DarKnesS_WolF: What about it? You didn't ask your original question AFTER I arrived. |
| 12:25.37 | psi0n | when i recieve a call on one trunk it appears to actually enter thru another trunk, both from the same provider. but the FROM_DID has the correct number. could this be causing any problems? |
| 12:25.37 | DarKnesS_WolF | [TK]D-Fender: got the idea? |
| 12:25.39 | [TK]D-Fender | DarKnesS_WolF: I'd have to see your actual file and code, and in your typical mannt, you have provided NEITHER to date. |
| 12:25.51 | DarKnesS_WolF | [TK]D-Fender: ok 1 min |
| 12:26.10 | psi0n | more specifically: -- Executing Set("SIP/xxxx2519-091b3af8", "FROM_DID=xxxx2501") in new stack |
| 12:26.35 | [TK]D-Fender | psi0n: You're the one who has to say if its causing probelms, and as for how you're configured or what you may have done inappropriately, we're not PSYCHIC. |
| 12:27.28 | psi0n | well, i am having a problem with setting DTMF to inband on the trunk, i just dont know if this could be related |
| 12:28.16 | DarKnesS_WolF | [TK]D-Fender: http://pastebin.ca/581199 |
| 12:28.18 | [TK]D-Fender | psi0n: No, thats DIALPLAN. Once the call is there your mode is already SET. |
| 12:28.45 | DarKnesS_WolF | the filename is passwords not password :-) it's typo from me in pasting |
| 12:29.08 | [TK]D-Fender | DarKnesS_WolF: Missing "s" <- and you don't need the "a" |
| 12:29.51 | DarKnesS_WolF | s ? |
| 12:29.55 | DarKnesS_WolF | in the options ? |
| 12:30.07 | [TK]D-Fender | DarKnesS_WolF: the typo you said was YOU, so nvm... |
| 12:30.25 | DarKnesS_WolF | ah okay no forget aout this s in passwords it was mistake but the a ? it said in the voip-info that it will not work unless i have the a option too |
| 12:30.41 | DarKnesS_WolF | but ok if i did remove the a it still didn't work |
| 12:30.44 | [TK]D-Fender | DarKnesS_WolF: pastebint he call attempt |
| 12:32.58 | *** join/#asterisk Corydon76-home (i=blue@pdpc/supporter/sustaining/Corydon76-home) |
| 12:32.58 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 12:33.51 | DarKnesS_WolF | [TK]D-Fender: i got only auth-incorrect " password incorrect " |
| 12:33.56 | DarKnesS_WolF | but give me 1 min i'll pastbin it |
| 12:34.47 | psi0n | ok then, to my actual problem: cell phones cant use my IVR, so my first guess is the DTMF has to be inband. however, setting it to inband doesnt seem to have any effect. |
| 12:35.36 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 12:35.36 | *** mode/#asterisk [+o anthm] by ChanServ |
| 12:38.06 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:40.47 | yonahw-work | is chan_zap.so not supposed to be in 1.4.5? |
| 12:42.26 | [TK]D-Fender | yonahw-work: Correct. |
| 12:42.54 | [TK]D-Fender | ~[TK]D-Fender |
| 12:42.55 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious. |
| 12:43.06 | cy303 | indeed |
| 12:43.06 | cy303 | heh |
| 12:43.08 | cy303 | lolz |
| 12:43.09 | tzanger | Captain obvious to the rescue! |
| 12:43.28 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 12:43.30 | yonahw-work | what replaces it? |
| 12:43.35 | waKKu | morning ;) |
| 12:45.57 | *** join/#asterisk bapril (n=bapril@pool-70-109-158-237.cncdnh.east.verizon.net) |
| 12:46.46 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 12:47.24 | [TK]D-Fender | yonahw-work: Nothing. I promise you there is no chan_zap in *1.4.5* |
| 12:47.53 | bapril | Anyone know if I want to record a MeetMe One Sip and One ZIP (Sometimes a second Zap), which is more efficient Monitor on the SIP leg, Monitor on the Zap leg or as an argument to the MeetMe? |
| 12:48.39 | [TK]D-Fender | file: i seem to recall that chan_zap being part of some OTHER tarball :) |
| 12:49.12 | file | zaptel itself is distributed as a separate thing, but chan_zap is part of Asterisk |
| 12:49.30 | [TK]D-Fender | file: Darn, well doesn't THAT just put a downer on my comedy! |
| 12:49.49 | file | no comedy allowed |
| 12:49.50 | [TK]D-Fender | yonahw-work: ok, scratch that. |
| 12:49.52 | yonahw-work | i am now thoroughly confused. |
| 12:49.59 | yonahw-work | oh so it is supposed to be there? |
| 12:50.17 | nexilus | yonahw-work: yes, if you make sure you compile zaptel/zapate BEFORE you compile asterisk |
| 12:50.46 | yonahw-work | was such, do i have to remove it from anywhere if upgrading from 1.2.x to 1.4.5? |
| 12:51.19 | JT | yonahw-work: did you compile zaptel first? |
| 12:51.23 | _VoiceMeUp_COM | i think you need |
| 12:51.57 | yonahw-work | oh actually come to think of it I may not have |
| 12:52.17 | yonahw-work | can i just compile zaptel, then compile asterisk or do i have to remove them first? |
| 12:52.25 | JT | that *may* be a problem |
| 12:52.28 | *** join/#asterisk Uatec (n=uatecuk@adsl.ntsols.com) |
| 12:52.30 | Uatec | Hello thar |
| 12:52.38 | flenders | JT: I updated the firmware and bootrom and now it doesn't even register |
| 12:52.39 | JT | probably safe to just recompile |
| 12:52.48 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 12:52.48 | *** mode/#asterisk [+o anthm] by ChanServ |
| 12:52.48 | JT | flenders: i see, you broke it! |
| 12:52.51 | Uatec | hey, has anybody heard when the next version of Asterisk Business Edition might be out? |
| 12:52.55 | flenders | JT: hahaha |
| 12:53.07 | s0ck | ~wikis |
| 12:53.08 | jbot | rumour has it, wikis is http://www.voip-info.org |
| 12:53.10 | flenders | JT: at least you can access the menus and stuff |
| 12:53.11 | yonahw-work | thanks for the tips |
| 12:53.41 | flenders | I installed bootrom 3.2.2 and sip 2.0.3 |
| 12:53.45 | DarKnesS_WolF | [TK]D-Fender: http://pastebin.ca/581246 |
| 12:53.51 | JT | flenders: well |
| 12:54.02 | JT | flenders: is it not freezing now? |
| 12:54.12 | JT | and it's loading sip? |
| 12:54.13 | flenders | no, not at all |
| 12:54.18 | flenders | yeah, apparently |
| 12:54.21 | JT | cool |
| 12:54.27 | JT | so it's just configured wrong, awesome |
| 12:54.34 | flenders | I can setup the lines and stuff, but it's not even trying to hit my asterisk |
| 12:54.40 | JT | it's a bit tricky at first |
| 12:54.46 | [TK]D-Fender | DarKnesS_WolF: PB "cat /etc/asterisk/password" and ""ls -l /etc/asterisk/" |
| 12:54.47 | JT | it's not how you'd expect |
| 12:54.57 | flenders | I can see that... and it takes sooo long to boot up |
| 12:55.41 | flenders | any tips? |
| 12:55.48 | JT | yeah, i've heard the newer firmwares boot slower |
| 12:55.52 | JT | so much stuff in it |
| 12:55.58 | DarKnesS_WolF | [TK]D-Fender: cat for file 110:1234 |
| 12:55.58 | DarKnesS_WolF | 108:4321 |
| 12:56.07 | flenders | it takes at least 5 minutes to boot up |
| 12:56.11 | JT | flenders: ouch |
| 12:56.17 | DarKnesS_WolF | [TK]D-Fender: -rw-r--r-- 1 root root 18 Jun 21 15:34 password |
| 12:56.23 | JT | flenders: ok, network configuration screen, leave at defaults |
| 12:56.50 | JT | Server 1 under SIP, put the server addess in |
| 12:56.53 | JT | address |
| 12:56.55 | flenders | I'm now using manual ip... manual and dhcp are working btw |
| 12:57.03 | JT | and register=1 |
| 12:57.46 | flenders | what about transport? |
| 12:57.50 | flenders | naptr? |
| 12:58.01 | JT | Lines > Line 1 > set Display Name, Address and Auth User ID to your sip user |
| 12:58.02 | flenders | register=yes |
| 12:58.10 | waKKu | folks.. have some way to reload configs of still connected peer on CLI ? |
| 12:58.13 | JT | Auth Password to sip password |
| 12:58.16 | [TK]D-Fender | DarKnesS_WolF: Dunno... |
| 12:58.16 | flenders | oh, web UI |
| 12:58.30 | JT | Server 1, address of SIP server |
| 12:58.45 | JT | the options are names slightly differently |
| 12:58.48 | JT | named |
| 12:58.56 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) [NETSPLIT VICTIM] |
| 12:58.56 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 12:58.56 | *** join/#asterisk mattfletcher (n=matt@88-97-179-134.dsl.zen.co.uk) [NETSPLIT VICTIM] |
| 12:58.56 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM] |
| 12:58.56 | *** join/#asterisk nitram (i=foo@superblob.com) [NETSPLIT VICTIM] |
| 12:58.56 | *** join/#asterisk deegan (i=deegan@killer.coding.ninja.monkii.net) [NETSPLIT VICTIM] |
| 12:58.59 | flenders | hangon |
| 13:00.17 | flenders | everytime I hit submit I have to wait 5 minutes to get on the UI again |
| 13:01.18 | DarKnesS_WolF | [TK]D-Fender: :-) thx |
| 13:01.25 | JT | nice |
| 13:01.45 | [TK]D-Fender | flenders: Using the Polycom Web-config? |
| 13:01.50 | flenders | yeah |
| 13:02.17 | flenders | sorry mate, first time on the polycoms |
| 13:02.19 | [TK]D-Fender | flenders: wrong answer! |
| 13:02.52 | flenders | what should I do then? |
| 13:02.56 | [TK]D-Fender | flenders: Get the SIP & BR images and provision them from scratch like you're supposed to. |
| 13:03.19 | flenders | I got sip 2.0.3 and bootrom 3.2.2 |
| 13:03.40 | flenders | but didn't know I was supposed to do it differently |
| 13:03.56 | flenders | should I edit the conf file? |
| 13:04.11 | [TK]D-Fender | flenders>I installed bootrom 3.2.2 and sip 2.0.3 <- old |
| 13:04.54 | [TK]D-Fender | flenders: General stuff goes into sip.cfg , you should have a phone[somenumber].cfg with reg specific (not even server IP in most cases) settings. |
| 13:05.04 | *** join/#asterisk MindTheGap (n=iote@c9503fb4.bhz.virtua.com.br) |
| 13:05.34 | flenders | where can I get the latest firmware? |
| 13:05.43 | MindTheGap | I have this as a macro: |
| 13:05.50 | JT | ~pb |
| 13:05.51 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 13:05.57 | MindTheGap | exten => s,1,Set(CDR(userfield)="FWDU - ${TIPO}") |
| 13:06.00 | MindTheGap | exten => s,2,Dial(sip/fwd/*1${MACRO_EXTEN:4},60,r) |
| 13:06.18 | MindTheGap | but CDR shows "s" as the dst... |
| 13:06.18 | Uatec | does anybody know how i can download an old version of misdn with conary, rather than the latest version? the latest version breaks my system |
| 13:06.36 | JT | Uatec: what card? |
| 13:06.41 | Uatec | B410P |
| 13:06.44 | Uatec | sorry |
| 13:06.44 | JT | oh |
| 13:06.48 | Uatec | mISDN-modules |
| 13:06.51 | Uatec | the card is irrelevant |
| 13:06.59 | MindTheGap | and i cannot set "dst" manually as it is read-only |
| 13:07.03 | Uatec | i just need to download the old misdn-modules |
| 13:07.10 | MindTheGap | how do I overcome this? |
| 13:07.57 | *** join/#asterisk dcm_ (n=dcm@207.59.3.77) |
| 13:07.58 | _VoiceMeUp_COM | http://www.tmz.com/tmz_main_video?titleid=987200446 |
| 13:08.06 | _VoiceMeUp_COM | man these rich jerks need more time in jail |
| 13:08.12 | _VoiceMeUp_COM | paris n brandon |
| 13:08.26 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 13:08.41 | JT | they only need more jail if they do something illegal, not because they're rich |
| 13:09.00 | _VoiceMeUp_COM | look at her leave the scene |
| 13:09.01 | Uatec | even just any way of downloading an old version of anything with conary, rather than the latest |
| 13:09.12 | _VoiceMeUp_COM | Reckless Driving plus illegal cut |
| 13:09.12 | _VoiceMeUp_COM | etc |
| 13:09.21 | waKKu | folks.. when edit something on features.conf ... how can i reload it ? |
| 13:09.24 | anonymouz666 | what's the correct behaviour when I call reachs a queue and then you transfer from A to B (both members) but B is busy? |
| 13:09.27 | _VoiceMeUp_COM | however you say that in english |
| 13:09.31 | anonymouz666 | should the caller listen MOH? |
| 13:09.35 | _VoiceMeUp_COM | Escalation ? |
| 13:10.05 | DrukenLPY | mmmm, my new office fridge works excelent :) |
| 13:11.05 | *** join/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 13:11.05 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:12.13 | *** join/#asterisk hackeron (n=hackeron@gentoo/user/hackeron) |
| 13:13.14 | yonahw-work | JT: I recompiled and chan_zap is on the list of modules not installed by this version should this concern me? |
| 13:13.32 | MindTheGap | I have this as a macro: |
| 13:13.33 | JT | if it's not installed, it's a problem |
| 13:13.36 | MindTheGap | exten => s,1,Set(CDR(userfield)="FWDU - ${TIPO}") |
| 13:13.37 | JT | MindTheGap: not again |
| 13:13.38 | MindTheGap | exten => s,2,Dial(sip/fwd/*1${MACRO_EXTEN:4},60,r) |
| 13:13.40 | *** join/#asterisk kombi_ (n=kombi@213.160.14.18) |
| 13:13.43 | JT | MindTheGap: saw it the first time |
| 13:13.47 | JT | MindTheGap: repeating is rude |
| 13:14.00 | hackeron | Hey, I moved from asterisk 1.2 to 1.4 and understandably everything is broken, lol -- it looks like the variables changed, like ${CALLERIDNUM} no longer works and it now goes to _0.-ANSWER,1 instead of _0.,1 etc etc -- is there like a little guide of configuration differences between the 2 versions? |
| 13:14.15 | yonahw-work | it was installed previously rathen than by this version even though I first compiled zaptel any suggestions? |
| 13:14.35 | MindTheGap | well JT, maybe you could answer then? |
| 13:14.47 | JT | MindTheGap: maybe you could be less demanding |
| 13:14.52 | JT | we are all here voluntarily |
| 13:14.55 | DrukenLPY | ya notice all the paris people are either hugely fat or trailer trash ugly? |
| 13:15.06 | kombi_ | can you run multiple softphones alongside each other on the same computer? (like for testing?) What is the prefered setup? |
| 13:15.29 | kombi_ | x-lite, idefisk etc.. |
| 13:15.35 | yonahw-work | kombi: yes |
| 13:15.38 | *** part/#asterisk andyd (n=andyd@host90-152-23-30.ipv4.regusnet.com) |
| 13:15.39 | yonahw-work | i have done it in windows |
| 13:15.43 | kombi_ | kewl! |
| 13:15.52 | JT | i sometimes run xpro, xlite and idefisk all at once |
| 13:16.01 | kombi_ | great! |
| 13:16.10 | JT | well, not quite at once |
| 13:16.14 | JT | but have them all running |
| 13:16.28 | JT | not talking on them all simultaneously |
| 13:16.31 | MindTheGap | Im aware of that, and Im not demanding... im asking, could you please answer? |
| 13:16.56 | JT | MindTheGap: have you considered the possibility that the lack of responses may be due to a lack of knowledge of the answer? |
| 13:17.00 | kombi_ | I am venturing into conferences today, so I need some kind of test setup.. |
| 13:17.06 | *** join/#asterisk grEvenX (n=even@ti500720a080-3214.bb.online.no) |
| 13:17.22 | JT | kombi_: sipp is a tool for load testing, too |
| 13:18.27 | kombi_ | JT: have got conferences running on your box? |
| 13:18.31 | MindTheGap | no I didnt... there are 269 ppl here... |
| 13:18.46 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:18.47 | JT | MindTheGap: and maybe 10 people actually paying attention, great |
| 13:19.01 | JT | MindTheGap: and demanding that *I* answer your question is a bit much |
| 13:19.04 | JT | kombi_: not much |
| 13:19.22 | mocker | Guh, to ask for a Mediatrix for work or to get another SPA. |
| 13:19.39 | mocker | We already have one Mediatrix.. |
| 13:19.54 | festr__ | hello, anyone seeing in 1.4 doubled CDR? |
| 13:20.06 | hackeron | Hmm, does _0. no longer work in asterisk 1.4? -- my outgoing plan seems to jump straight to _0.-ANSWER,1 --- any ideas? |
| 13:20.46 | DrukenLPY | JT: you da man... hehe |
| 13:20.47 | MindTheGap | as I said , im not demanding, im politely asking for help, not yours anymore by the way... |
| 13:20.53 | nexilus | hackeron: have apastebin of the dialplan? |
| 13:22.10 | [TK]D-Fender | hackeron: the old CALLERID vares are GONE |
| 13:22.48 | [TK]D-Fender | hackeron: They were deprecated in *1.2* and you never read EITHER of the upgrade.txt's or dozens of articles and code samples that should have tipped you off. |
| 13:22.49 | hackeron | nexilus: sure: http://rafb.net/p/HmSlQW52.html |
| 13:23.09 | [TK]D-Fender | BAD : exten => _0.,7,SetCallerID(02070993461) |
| 13:23.40 | hackeron | [TK]D-Fender: hmm, so what do I use instead? |
| 13:24.10 | [TK]D-Fender | hackeron: "show function CALLERID" |
| 13:24.10 | hackeron | [TK]D-Fender: but thing is it doesn't even show Outgoing call to ... -- it just goes straight to _0.-ANSWER |
| 13:24.57 | hackeron | [TK]D-Fender: The 'show function' command is deprecated and will be removed |
| 13:25.13 | [TK]D-Fender | hackeron: "core show function CALLERID" I believe now. |
| 13:25.52 | [TK]D-Fender | hackeron: And I don't think its appropriate for you to put stuff after the "." in your pattern match. |
| 13:26.03 | hackeron | ok, so that's CALLERID(num,02070993461) now? |
| 13:26.12 | [TK]D-Fender | hackeron: Also a little shocked if LookupCIDName still exists |
| 13:26.27 | hackeron | [TK]D-Fender: still there but does show deprecated errors now |
| 13:26.32 | [TK]D-Fender | hackeron: No. Go read the instructions, and go read up on "asterisk variables" on the WIKI. |
| 13:26.42 | hackeron | [TK]D-Fender: voip-info.org? |
| 13:26.46 | [TK]D-Fender | hackeron: yes |
| 13:26.52 | nexilus | hackeron: i use Set(CALLERID(num)=sumthn) |
| 13:26.53 | hackeron | [TK]D-Fender: well, that's what I'm reading |
| 13:26.59 | hackeron | [TK]D-Fender: that's where I got all the stuff from |
| 13:27.13 | [TK]D-Fender | Wrong style : exten => _0.-ANSWER,1,NoOp(HACKERON: Successful Call) |
| 13:27.30 | mocker | [TK]D-Fender: You ever tried fax on one of those SPA-2102s? |
| 13:27.36 | s0ck | P[ 1] * IND : HANGUP pid:20 ctx:Intern dad:746235 oad:01554741144 State:EXTCANTMATCH |
| 13:27.45 | mocker | (Fax would be coming from a PRI initially, not IP) |
| 13:27.46 | s0ck | isdn2 (bri) |
| 13:27.53 | s0ck | dad=did? |
| 13:27.55 | [TK]D-Fender | ^^^ don't just slap a suffix on that original pattern match, jump to something sane & fixed |
| 13:27.57 | hackeron | [TK]D-Fender: well, what should it be instead? -- that worked with 1.2 |
| 13:28.37 | [TK]D-Fender | hackeron: something like "RESULT-${DIALSTATUS}" with no underscore, etc. |
| 13:29.05 | *** join/#asterisk yonahw-work (n=yonahw-w@genie03-173-74.inter.net.il) |
| 13:29.28 | hackeron | [TK]D-Fender: errr, but I want to override default falback -- say if the call is cancelled it normally does autofallthrough, but it says on the wiki if I add -CANCEL for example, then I can override |
| 13:30.05 | [TK]D-Fender | hackeron: What default fallback?! |
| 13:30.49 | hackeron | [TK]D-Fender: ? |
| 13:31.53 | hackeron | [TK]D-Fender: what I don'g get is why does asterisk go to _0.-ANSWER by default? |
| 13:32.29 | [TK]D-Fender | hackeron: because you have clearly not read the instructions for DIAL. |
| 13:32.40 | hackeron | [TK]D-Fender: all the stuff I have for _0., is completely ignored -- when I try to dial out, the first thing it does is _0.-ANSWER,1,NoOp(HACKERON: Successful Call) |
| 13:33.03 | flenders | alright! polycom now registers! |
| 13:33.17 | [TK]D-Fender | hackeron: UNDO all those semi-pattern-matched JUMP POINTS, and make them FIXED |
| 13:34.32 | hackeron | [TK]D-Fender: I'm reading the wiki now for the Dial command, all the information there is for asterisk 1.2 -- is there any asterisk 1.4 documentation? |
| 13:34.55 | [TK]D-Fender | hackeron: "core show application dial" <- |
| 13:34.59 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.219.nw.nuvox.net) |
| 13:35.21 | *** join/#asterisk mosty (n=mostyn@60-241-198-194.static.tpgi.com.au) |
| 13:35.49 | [TK]D-Fender | hackeron: exten => _0.,5,SetVar(CALLFILENAME=${TIMESTAMP}-OUT-${EXTEN}) <- Setvar... GONE! Welcome "Set" |
| 13:35.50 | hackeron | [TK]D-Fender: I don't get it, so what do I do if I want to jump to ${DIALSTATUS} ? |
| 13:36.19 | [TK]D-Fender | hackeron: exten => _0.-ANSWER,1,NoOp(HACKERON: Successful Call) <- bad. |
| 13:36.27 | hackeron | [TK]D-Fender: why? |
| 13:36.29 | [TK]D-Fender | hackeron: exten => STATUS-ANSWER,1,NoOp(HACKERON: Successful Call) <- better |
| 13:36.44 | [TK]D-Fender | hackeron: It should not be a pattern match. |
| 13:37.05 | [TK]D-Fender | hackeron: The mere fact you are doing all of that on a numbered extension in the first place is BAD. |
| 13:37.08 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 13:37.23 | [TK]D-Fender | hackeron: Thats why these sort of things are done in macros. |
| 13:37.34 | hackeron | [TK]D-Fender: err, I want to cover all outgoing calls |
| 13:38.05 | hackeron | [TK]D-Fender: so I need to change to exten => _0.,9,Goto(OUTGOING-${DIALSTATUS},1) ?? |
| 13:38.14 | [TK]D-Fender | hackeron: thats fine, and irrelevant. Dump all that in a macro and have _0. call it. |
| 13:38.44 | [TK]D-Fender | hackeron: Sure, thats much better. make sure the landing points match. |
| 13:39.19 | *** join/#asterisk coppice (n=chatzill@10.198.17.210.dyn.pacific.net.hk) |
| 13:39.23 | *** join/#asterisk deeperror (n=deeperro@69-215-202-202.ded.ameritech.net) |
| 13:40.10 | deeperror | Is it possible to capture errors when creating .call files from a script? |
| 13:40.15 | waKKu | folks.. need some help with pickupgroups ... i edit features.conf and set key to *2, i put a group of users on the same callgroup and pickupgroup .... but, when i try to pickupgroup with *2 i get message on CLI: Jun 21 10:36:58 NOTICE[4828]: chan_iax2.c:7198 socket_read: Rejected connect attempt from 201.56.112.192, request '*2@internacional' does not exist |
| 13:40.48 | waKKu | i did reload on features and register users again |
| 13:41.50 | *** join/#asterisk ELBunce (n=erik@kde/developer/bunce) |
| 13:43.16 | s0ck | http://forums.digium.com/viewtopic.php?t=16307&highlight=misdn |
| 13:43.22 | s0ck | therein lies the problem, it seems |
| 13:43.58 | *** join/#asterisk lilalinux (i=e-trolle@langweiligneutral.deswahnsinns.de) |
| 13:44.22 | hackeron | [TK]D-Fender: OK, I replaced setvar with set and the _0.-STUFF with OUTGOING-STUFF -- seems to work now -- I'll go read the changelogs now from the last version I used this config and make more of a sane config |
| 13:44.28 | hackeron | [TK]D-Fender: thanks very much for all your help |
| 13:44.49 | [TK]D-Fender | hackeron: keep it up! |
| 13:47.21 | lilalinux | I had 1 hfc card (zaptel) in my system for NT mode and everyting works so far with: "span=1,1,3,ccs,ami ; bchan=1-2 ; dchan=3". Now I have a 2nd hfc card for TE mode, how do I need to change /etc/zaptel.conf? |
| 13:47.27 | *** join/#asterisk seva (i=seva@sevatech.com) |
| 13:47.35 | seva | is there a way to adjust gain on a sip channel? |
| 13:47.36 | *** join/#asterisk dikdust (n=dikdust@gandalf.ipv6.adfacom.it) |
| 13:47.48 | seva | or iax for that matter |
| 13:47.49 | JT | lilalinux: bristuff? |
| 13:47.55 | lilalinux | JT: yes |
| 13:48.19 | JT | span=1,0,0, etc |
| 13:48.27 | JT | not 1,1,3 |
| 13:48.31 | [TK]D-Fender | seva No. |
| 13:48.32 | JT | for NT |
| 13:48.43 | lilalinux | JT: k |
| 13:48.46 | lilalinux | and for TE? |
| 13:48.51 | JT | you need to add a new span for TE |
| 13:49.01 | JT | 2,1,0 |
| 13:49.17 | lilalinux | and bchan&dchan? |
| 13:49.27 | JT | well |
| 13:49.33 | JT | that's simply a matter of counting :) |
| 13:49.38 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 13:49.42 | JT | what's after 3? |
| 13:49.50 | lilalinux | :^) |
| 13:49.51 | rob0 | um ... 5? |
| 13:50.04 | lilalinux | thx |
| 13:50.13 | lilalinux | ,css,ami is ok? |
| 13:50.16 | JT | yes |
| 13:50.18 | tzafrir | Feedback welcome: |
| 13:50.18 | lilalinux | thx |
| 13:50.21 | tzafrir | http://svn.digium.com/svn/zaptel/branches/1.4/README |
| 13:50.24 | JT | well |
| 13:50.41 | tzafrir | lilalinux, for BRI: yes |
| 13:50.52 | rob0 | Three is the number to which thou shalt count, and the number that thou shalt count shall be three. |
| 13:50.57 | tzafrir | lilalinux, for zapbri, use an up-to-date genzaptelconf... |
| 13:51.11 | *** join/#asterisk ghento (n=ghento@CPE0014bfaf7d7a-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 13:51.30 | JT | lilalinux: you need to modify zapata.conf too |
| 13:51.37 | _VoiceMeUp_COM | hmm |
| 13:51.48 | seva | [TK]D-Fender: what are the options if people sounds too quiet? |
| 13:51.48 | _VoiceMeUp_COM | Anyone here know anything about 011880 scams ? |
| 13:51.54 | tzafrir | also feedback for: |
| 13:52.14 | mosty | _VoiceMeUp_COM, is that a toll number prefix? |
| 13:52.16 | tzafrir | http://updates.xorcom.com/astribank/bristuff/INSTALL.html |
| 13:52.16 | lilalinux | TE mode doesn't need ,ccs,ami? |
| 13:52.21 | _VoiceMeUp_COM | nope |
| 13:52.22 | [TK]D-Fender | seva : You need to fix it at the SOURCE. |
| 13:52.26 | _VoiceMeUp_COM | at 6$per minute i doubt |
| 13:52.30 | _VoiceMeUp_COM | Bangladesh |
| 13:52.54 | seva | [TK]D-Fender: any suggestions on how to increase source volume for a headset? |
| 13:52.54 | mosty | _VoiceMeUp_COM, what is it? |
| 13:53.17 | JT | lilalinux: yes it does |
| 13:53.26 | [TK]D-Fender | seva : Look at whatever its plugged INTO. |
| 13:53.39 | *** join/#asterisk docelmo (n=vircuser@c-76-99-157-112.hsd1.de.comcast.net) |
| 13:53.50 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
| 13:54.14 | _VoiceMeUp_COM | not sure yet |
| 13:54.54 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 13:55.06 | *** join/#asterisk elriah (i=elriah@175.sub-75-201-96.myvzw.com) |
| 13:56.38 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 13:57.14 | lilalinux | JT: please forgive me, but your answer is ambiguous |
| 13:59.04 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 14:00.56 | *** join/#asterisk holiday_42 (n=no@spike.wcta.net) |
| 14:01.14 | *** join/#asterisk grandpapadot (i=elriah@82.sub-75-200-91.myvzw.com) |
| 14:01.21 | JT | lilalinux: yes, of course it needs all that stuff |
| 14:01.37 | JT | lilalinux: you know there are sample configs that come with bristuff, right? |
| 14:01.45 | JT | and zaptel.conf is documented |
| 14:01.56 | grandpapadot | Hi all. Suddently, it seems out of nowhere, we're having weird caller id issues with 1.2.18. Caller ID's are just wrong in some cases and in other cases it seems channels are getting caller id info from other channels. Any suggestions? |
| 14:02.05 | lilalinux | JT: sry |
| 14:03.58 | _VoiceMeUp_COM | grandpapadot that is weird |
| 14:04.14 | deeperror | how can i get more debug information from the manager? |
| 14:05.54 | waKKu | someone there using pickupgroups ? ... need set sth on extensions.conf ? |
| 14:06.03 | Uatec | hi, i've got a BRI isdn line coming in to my Asterisk box, with two lines in it, obviously |
| 14:06.09 | Uatec | i can receive two incomming calls at a time |
| 14:06.23 | Uatec | and i can make 1 out bound call |
| 14:06.33 | Uatec | but when i try to make a second concurrent out bound call, i get Congestion!! |
| 14:06.36 | Uatec | what's that about? |
| 14:07.16 | mosty | Uatec, what's your dial command look like? |
| 14:08.08 | Uatec | mosty Dial( |
| 14:08.12 | Uatec | exten => _9.,n,Dial(mISDN/g:isdn/${EXTEN:1}) |
| 14:08.30 | *** join/#asterisk _DAW_ (n=chatzill@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:08.53 | Uatec | waKKu, you need to set that on a per channel basis. i.e. in sip.conf |
| 14:09.27 | waKKu | Uatec i did it.. r u saying about pickupgroup and callgroup, right ? |
| 14:09.37 | Uatec | yes |
| 14:09.40 | waKKu | i'm using 3 peers iax |
| 14:09.44 | mosty | Uatec, looks ok |
| 14:10.42 | waKKu | Uatec hm... i'm using idefisk... maybe a problem ? |
| 14:12.22 | *** join/#asterisk seele_ (n=seele@dns.datawareltda.com) |
| 14:12.28 | seele_ | hello |
| 14:12.37 | Uatec | shouldn't make any difference waKKu |
| 14:13.13 | waKKu | yeah.. i think so too ;/... but still have problem |
| 14:13.36 | festr__ | anyone seeing in 1.4 doubled CDR records when one call is hangup in ring state? |
| 14:14.09 | Uatec | when i try to make a call i get "No free channel in group XXXXXXX!" |
| 14:14.16 | Uatec | where XXXXXXX appears to be a completely random number |
| 14:15.00 | *** join/#asterisk _DAW_ (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:15.05 | Uatec | usually in the region of 1309115312 |
| 14:15.06 | Uatec | though |
| 14:16.13 | seele_ | someone can help me with always-on agent login |
| 14:18.32 | jkiff | seele_: Not until you describe your problem. ;) |
| 14:18.45 | MindTheGap | I have this as a macro: |
| 14:18.51 | MindTheGap | exten => s,1,Set(CDR(userfield)="FWDU - ${TIPO}") |
| 14:18.58 | MindTheGap | exten => s,2,Dial(sip/fwd/*1${MACRO_EXTEN:4},60,r) |
| 14:19.04 | MindTheGap | CDR shows "s" as the dst and I cand do a Set(CDR(dst)=$MACRO_EXTEN) as the dst var is readonly. |
| 14:19.10 | MindTheGap | how do I overcome that? |
| 14:19.35 | Mercestes | seele_, Isn't "always-on" and "login" an oxymoron? |
| 14:19.55 | *** join/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 14:20.10 | Mercestes | seele_, queues.conf: member => SIP/1234 Like that? |
| 14:20.11 | seele_ | An always-on agent will sit all day with his headphones on, listening to music, until a |
| 14:20.11 | seele_ | call comes in |
| 14:20.31 | Mercestes | and then what happens? |
| 14:20.52 | seele_ | Mercestes, how can i make it |
| 14:21.02 | [TK]D-Fender | Mercestes: I think they ditched the "tie the agent up on an actual call Queue App" |
| 14:21.08 | *** part/#asterisk NoCarrier (n=NoCarrie@unaffiliated/badpacket) |
| 14:21.19 | [TK]D-Fender | Mercestes: Few people used it. |
| 14:21.23 | seele_ | I have this login method exten => 299,1,AgentCallBackLogin(|@from-internal) |
| 14:21.23 | seele_ | exten => 298,1,AgentLogin() |
| 14:21.28 | robl^ | hrmmm.. should I dare to try SLA in 1.4.x? |
| 14:22.36 | seele_ | I have 2 asterisk in different places and I need to register agents from the second asterisk to a queue in the first asterisk |
| 14:22.49 | [TK]D-Fender | seele_: Hrm... seems to be there still, so 298 should work |
| 14:23.01 | seele_ | nop |
| 14:23.01 | *** join/#asterisk syneus (n=syneus@81.88.246.130) |
| 14:23.21 | seele_ | because the second asterisk has a sip trunk with the first asterisk |
| 14:23.34 | seele_ | and all the calls are routed to other extension |
| 14:23.43 | [TK]D-Fender | seele_: that makes NO sense. That app has NOTHING to do with SIP or other servers. |
| 14:23.54 | seele_ | when the first make a callback |
| 14:24.13 | seele_ | example: |
| 14:24.14 | [TK]D-Fender | seele_: AgentLogin has NOTHING to do with AgentCallbackLogin. |
| 14:24.37 | seele_ | I'm in the second place and I need to login into a queue in the first place |
| 14:24.47 | seele_ | I dial 298 |
| 14:25.29 | *** join/#asterisk hfb (n=hfb@pool-72-87-254-188.lsanca.dsl-w.verizon.net) |
| 14:25.41 | *** part/#asterisk seva (i=seva@sevatech.com) |
| 14:25.45 | seele_ | and the first place asterisk answer ...I'm logon but the calls no go to the second place |
| 14:25.54 | Mercestes | seele_, Oh, I think a combination of musiconhold, and SipAddHeader and AUTO_ANSWER and queues. |
| 14:26.53 | [TK]D-Fender | seele_: 298 asks you to enter your agent # and you just SIT THERE waiting for a beep, not a CALL. |
| 14:27.23 | [TK]D-Fender | seele_: you don't seem to understand AgentLogin's purpose at all. |
| 14:28.14 | seele_ | no, 298 ask my agent number my password an then I hang then the asterisk calls when a queue receives a call |
| 14:28.56 | seele_ | [TK]D-Fender, I need the beep one registration |
| 14:29.06 | seele_ | no the callback registration |
| 14:29.44 | [TK]D-Fender | seele_: NO. You are NOT supoosed to hang up after doing AgentLogin! |
| 14:30.10 | [TK]D-Fender | seele_: You SIT THERE ON THE CALL and WAIT for the beep. it does NOT call you back. |
| 14:30.16 | seele_ | with the 298 yes .... is callback login method |
| 14:30.21 | *** join/#asterisk rmayorga (i=rmyorg@unaffiliated/rmayorga) |
| 14:30.23 | Uatec | i get the error, No free channel in group... but the only thing google brings up about that is it's listing in the source... |
| 14:30.24 | Uatec | what use is that? |
| 14:30.26 | Uatec | what does it mean |
| 14:30.41 | JT | maybe your group is setup wrong |
| 14:30.53 | Uatec | maybe.. |
| 14:31.03 | [TK]D-Fender | seele_>exten => 298,1,AgentLogin() <- This. Is. NOT. a. CALLBACK |
| 14:31.13 | Uatec | but how? |
| 14:31.22 | seele_ | hmmm |
| 14:31.35 | JT | Uatec: so far i've yet to see you pastebin anything like your config |
| 14:31.46 | JT | Uatec: you're wasting your time until you do :) |
| 14:31.48 | [TK]D-Fender | seele_: PUT DOWN THE CRACK-PIPE! (c) JerJer |
| 14:32.43 | Uatec | http://rafb.net/p/EVCf7C12.html like that? |
| 14:32.55 | Nugget | http://www.liewcf.com/blog/wp-images/ikea.jpg <-- heh |
| 14:33.09 | Uatec | and my dial appears in the CLI as: Dial("SIP/recsdesk-0825a830", "mISDN/g:isdn/123") |
| 14:33.38 | *** join/#asterisk allen__s (n=chatzill@72.242.225.99) |
| 14:33.42 | jkiff | seele_: Do a 'show application' for AgentLogin and AgentCallbackLogin on the CLI. |
| 14:34.03 | jkiff | The wisdom and power will flow through your veins. |
| 14:35.15 | seele_ | sorry my error |
| 14:35.21 | seele_ | this si my actual app |
| 14:35.21 | seele_ | exten=> 451,1,AgentCallbackLogin(||${CALLERIDNUM}@from-internal) |
| 14:35.21 | seele_ | exten=> 450,1,AgentCallbackLogin(||l) |
| 14:35.28 | *** join/#asterisk allen__s (n=chatzill@72.242.225.99) |
| 14:39.00 | kombi_ | Maybe I got this slightly wrong: To be able to invite a third party into a call, do I use meetme()? |
| 14:39.11 | JT | Uatec: how many bris do you have? |
| 14:39.41 | Mercestes | kombi_, if memory serves |
| 14:39.46 | *** join/#asterisk syneus (n=syneus@81.88.252.94) |
| 14:40.00 | Uatec | jt, just the one |
| 14:40.20 | JT | Uatec: seeing the possible problem?: |
| 14:40.21 | JT | [isdn] |
| 14:40.22 | JT | ports=1,2,3,4 |
| 14:40.39 | Uatec | it should still cycle and use the first available channel |
| 14:40.42 | JT | dialling on non existant channel usually causes problems |
| 14:40.45 | JT | not true |
| 14:40.57 | Uatec | not? |
| 14:41.02 | Uatec | what would it dial one? |
| 14:41.03 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 14:41.07 | JT | define ports that exists |
| 14:41.10 | JT | not ones that don't |
| 14:41.11 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 14:41.59 | Uatec | ok |
| 14:42.10 | Uatec | well i've set it to use the only port i have plugged in at the moment |
| 14:42.16 | Uatec | it still doesnt' work |
| 14:42.30 | waKKu | folks.. whats more i need to do to use pickups for a group ? 1) i set same pickupgroup and callgroup on iax.conf for 4 peers ... 2) Config features.conf to pickupexten => *5 - reload res_features.so; iax2 reload .... still doesnt work.. someone can help me ? |
| 14:42.35 | Uatec | http://rafb.net/p/jocNJp84.html |
| 14:42.38 | Uatec | like that |
| 14:42.42 | Uatec | it says that port 2 is already in use |
| 14:42.47 | Uatec | but there are two lines on each port... |
| 14:43.03 | *** join/#asterisk unixlike (n=dsadas@31.67.modemcable.oricom.ca) |
| 14:43.19 | JT | right |
| 14:43.22 | JT | not terribly sure |
| 14:43.27 | JT | personally i avoid misdn |
| 14:43.31 | JT | it makes my head hurt |
| 14:43.42 | Uatec | lol |
| 14:43.51 | Uatec | what would you use to connect to a BRI instead? |
| 14:44.01 | unixlike | is it possible to register more than 1 iax account at the same time in iax.conf ? |
| 14:44.26 | mosty | yes |
| 14:44.35 | JT | bristuff |
| 14:45.25 | Uatec | they're designed for the junghanns hardware though |
| 14:45.39 | JT | it works fine with plenty of cards |
| 14:45.41 | JT | beronet |
| 14:45.48 | Uatec | and i'm using a B410P, and using digiums suggested, i.e. misdn |
| 14:45.49 | JT | all the single port hfc-s passive cards too |
| 14:46.09 | JT | yeah i wouldn't buy a bri card from digium |
| 14:46.20 | JT | but it may be driver compatible with bristuff's qozap |
| 14:46.22 | JT | dunno |
| 14:46.26 | Uatec | well i've got it now |
| 14:46.29 | Uatec | and it cost £5000 |
| 14:46.30 | Uatec | -0 |
| 14:46.32 | Uatec | lol |
| 14:46.37 | *** join/#asterisk forsaken_ (n=puga@ns1.erimat.com.br) |
| 14:46.40 | JT | 5000 pounds, are you joking? |
| 14:46.47 | Uatec | 500 |
| 14:46.51 | Uatec | typo |
| 14:47.03 | JT | a little better |
| 14:47.18 | forsaken_ | good morning (for those who lives in the same timezone that me) |
| 14:47.23 | JT | the junghanns card is cheaper |
| 14:47.27 | waKKu | morning ;) |
| 14:47.34 | waKKu | brazil says |
| 14:47.56 | Uatec | the b410p has all sorts of things built in, like echo cancellation apparently |
| 14:48.04 | Uatec | my boss reckons it's a worth while purchase |
| 14:48.17 | JT | and it's designed in a country where you can't even buy eurobri isdn |
| 14:48.21 | Uatec | and he would fire me, before replacing it for any reason other than it's physically broken |
| 14:48.26 | JT | so good luck with support :) |
| 14:48.32 | Uatec | lol |
| 14:48.33 | Uatec | i know |
| 14:48.37 | Uatec | support is shit for it |
| 14:49.54 | JT | i haven't heard of anyone trying bristuff with it |
| 14:49.56 | s0ck | i had an issue with a b410p earlier |
| 14:50.01 | s0ck | exten would not pass calls out |
| 14:50.02 | JT | but i am interested if it works |
| 14:50.19 | s0ck | $OUTNUM$ worked |
| 14:50.35 | *** join/#asterisk joetester (n=joeteste@216.191.34.13) |
| 14:50.35 | JT | misdn drives me batty, it's alpha quality software |
| 14:50.42 | s0ck | this is the first one im setting up |
| 14:50.48 | s0ck | still wrestling with inbound calls not working |
| 14:51.00 | Uatec | s0ck |
| 14:51.04 | Mercestes | JT: how do you *really* feel about it? |
| 14:51.05 | Uatec | are you using bristuff of misdn? |
| 14:51.09 | s0ck | misdn |
| 14:51.13 | Uatec | ok |
| 14:51.13 | s0ck | as per digium docs |
| 14:51.21 | Uatec | well i've personally never had any problem with incoming calls |
| 14:51.25 | Uatec | after i got the thing working at all, that is |
| 14:51.26 | JT | bristuff is much closer to the asterisk way doing things, but digium recommend misdn for political reasons i think |
| 14:51.26 | Uatec | :P |
| 14:51.32 | s0ck | what kernel you running? |
| 14:51.42 | Uatec | 2.6.17 |
| 14:51.47 | Uatec | 2.6.19 screwed it up completely |
| 14:51.50 | s0ck | anything above .15 allegedly works |
| 14:51.51 | Uatec | broke misdn and zapata |
| 14:52.08 | s0ck | which doesn't help me running .9 :P |
| 14:52.16 | s0ck | recompiling now. |
| 14:52.25 | Uatec | which distro are you using? |
| 14:52.28 | JT | just a tip, never ever try to run misdn in NT mode |
| 14:52.44 | s0ck | centos |
| 14:53.01 | Uatec | ahh |
| 14:53.02 | s0ck | normally use slack |
| 14:53.07 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 14:53.07 | forsaken_ | I would like to know: if I use the same character in the blindxfer and atxfer like #, will it work?? |
| 14:53.08 | Uatec | i'm using PoundKeyLinux |
| 14:53.09 | s0ck | thought i'd try something different |
| 14:53.21 | mosty | forsaken_, only one of them can work |
| 14:53.23 | Uatec | s0ck, what msns are you setup to use? |
| 14:53.33 | s0ck | it has this cool command 'yum upgrade kernel-smp' |
| 14:53.34 | mosty | forsaken_, asterisk can't magically guess which one you want |
| 14:53.41 | s0ck | great! i though |
| 14:53.41 | s0ck | t |
| 14:53.54 | *** join/#asterisk wunderkin (i=wunderki@ip68-104-149-97.ph.ph.cox.net) |
| 14:53.55 | s0ck | it updated to some minor revision and would go no further |
| 14:54.05 | forsaken_ | mosty, hehehe sorry and tks =P |
| 14:54.07 | Mercestes | forsaken_, load up chan_telepathy.so |
| 14:54.16 | forsaken_ | Mercestes u.u` |
| 14:54.46 | Uatec | JT, what is NT mode? |
| 14:54.54 | JT | acting as exchange |
| 14:55.04 | JT | connecting ISDN phones of PABXes to it |
| 14:55.07 | JT | s/of/or/ |
| 14:55.21 | JT | Network Terminator |
| 14:55.26 | JT | as opposed to Terminal Equipment |
| 14:56.00 | Uatec | ah |
| 14:56.07 | s0ck | i thought it was one or the other |
| 14:56.35 | *** part/#asterisk deeperror (n=deeperro@69-215-202-202.ded.ameritech.net) |
| 14:56.54 | [TK]D-Fender | kombi_>Maybe I got this slightly wrong: To be able to invite a third party into a call, do I use meetme()? <- no |
| 14:57.03 | s0ck | nt=handset |
| 14:57.05 | s0ck | te=pbx |
| 14:57.06 | s0ck | or not? |
| 14:57.08 | [TK]D-Fender | kombi_: You use whatever 3-way calling feature your phone aready support. |
| 14:57.16 | [TK]D-Fender | s0ck: Correct |
| 14:57.33 | *** join/#asterisk [GuS] (n=gdnet@unaffiliated/gus/x-663402) |
| 14:57.43 | JT | s0ck: no |
| 14:57.50 | JT | s0ck: it's as i explained |
| 14:57.58 | JT | NT acts as exchange |
| 14:58.03 | [GuS] | Hi Guys!... i using latest Asterisk server, and i wanted to ask... does support SIMPLE? |
| 14:58.05 | JT | TE acts as terminal |
| 14:58.08 | [GuS] | for chat |
| 14:58.20 | kombi_ | [TK]D-Fender: so you mean our beloved * isn't necessarily involved when making a three way call? |
| 14:58.22 | [GuS] | (SIP/SIMPLE) |
| 14:58.55 | [TK]D-Fender | kombi_: Sort of. In the case of your typicaly SIP phone, the phone places the 2nd call and IT does the bridging. |
| 14:59.16 | [TK]D-Fender | [GuS]: * does not support SIP Messaging |
| 14:59.28 | kombi_ | ..thus making the job easier here, splendid! |
| 14:59.33 | [GuS] | ok, thanks |
| 14:59.46 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 15:00.13 | [GuS] | [TK]D-Fender: it is planned on development? |
| 15:00.52 | [TK]D-Fender | ~8ball any SIP messaging support on its way? |
| 15:00.52 | jbot | Are you smoking crack? |
| 15:00.55 | [TK]D-Fender | :O |
| 15:01.11 | holiday_42 | lol |
| 15:01.15 | [GuS] | ? |
| 15:01.24 | [TK]D-Fender | [GuS]: Translation : NO |
| 15:01.37 | rob0 | [GuS] is the new SIP messaging development director. |
| 15:01.44 | [GuS] | [TK]D-Fender: you always ironic :P |
| 15:02.12 | [TK]D-Fender | [GuS]: Don't you think? |
| 15:02.37 | [GuS] | for sure! |
| 15:02.53 | MindTheGap | I have this as a macro: |
| 15:02.58 | MindTheGap | exten => s,1,Set(CDR(userfield)="FWDU - ${TIPO}") |
| 15:02.58 | *** join/#asterisk ghento (n=ghento@bas8-toronto01-1279270360.dsl.bell.ca) |
| 15:03.03 | MindTheGap | exten => s,2,Dial(sip/fwd/*1${MACRO_EXTEN:4},60,r) |
| 15:03.09 | MindTheGap | CDR shows "s" as the dst and I cand do a Set(CDR(dst)=$MACRO_EXTEN) as the dst var is readonly. |
| 15:03.15 | MindTheGap | how do I overcome that? |
| 15:03.42 | MindTheGap | if not using macros, the CDR is ok... |
| 15:03.59 | JT | maybe you should ask the question a few dozen more times, MindTheGap |
| 15:04.05 | MindTheGap | but it uses macro s prio as dst... weird... |
| 15:04.55 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:05.00 | *** join/#asterisk phearless (n=phear@host217-34-75-65.in-addr.btopenworld.com) |
| 15:05.46 | [TK]D-Fender | MindTheGap: Pastebin ALL of the diaplan in this call's path from beginning to end, including CLI out, and stop repeating it over & over. If you don't gen an answer wait a few hours or try asking on the mailing lists. |
| 15:09.57 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 15:10.23 | [GuS] | seems i found a way :P http://www.voip-info.org/wiki/view/Asterisk+SIP+Messaging |
| 15:11.40 | *** part/#asterisk dcm_ (n=dcm@207.59.3.77) |
| 15:11.43 | MindTheGap | damn, would someone please drop in a few beers and a naked woman in this channel? ppl are so stressed today... |
| 15:12.07 | JT | MindTheGap: looks like you're new here |
| 15:12.25 | [TK]D-Fender | [GuS]: "Wed, 17 Aug 2005 09:37:19 -0700" <------------ |
| 15:12.32 | MindTheGap | not exactly JT... |
| 15:12.55 | JT | MindTheGap: i didn't see anything unreasonable in what [TK]D-Fender requested |
| 15:13.05 | JT | very standard here |
| 15:13.34 | *** join/#asterisk zeeesh (i=zeeesh@14-237-154-202.wol.net.pk) |
| 15:13.35 | zeeesh | hi |
| 15:13.46 | [GuS] | * Last modification by Lukas Oberhuber on Tue 02 of Jan, 2007 [22:01] |
| 15:15.36 | [TK]D-Fender | [GuS]: Ok, if you've found a version that can be adapted to today's code, more power to you. Keep in mind 1.2 was barely out. I'd be VERY surprised of it fits in 1.4 |
| 15:15.56 | [GuS] | yeah i know |
| 15:16.11 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 15:16.11 | [GuS] | but i least i could see how he done it |
| 15:16.58 | robl^ | MindTheGap: (and a few naked men for some of the members) ;-) |
| 15:17.10 | MindTheGap | neither did I, as standard as unessesary comments nobody complains about... it just that today is seems a bit more... |
| 15:17.16 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-211-202.dsl.irvnca.pacbell.net) |
| 15:17.24 | *** join/#asterisk brea (n=brea@c-71-195-248-169.hsd1.ma.comcast.net) |
| 15:17.34 | MindTheGap | see what i mean? |
| 15:17.40 | JT | what unnecessary comments? |
| 15:17.43 | JT | not particularly |
| 15:18.02 | MindTheGap | never mind... peace... |
| 15:19.07 | MindTheGap | and yes, mea culpa, i know repeating anoying... |
| 15:23.33 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 15:25.13 | *** join/#asterisk apardo (n=deal@49.145.217.87.dynamic.jazztel.es) |
| 15:25.37 | *** join/#asterisk ghento (n=ghento@76.64.29.216) |
| 15:27.14 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 15:28.37 | [jwb] | does Read() have some sort of additional sound file format/location restrictions as compared to Playback() ? |
| 15:28.44 | [jwb] | (under 1.2) |
| 15:29.02 | *** join/#asterisk boch (n=fran@190.48.195.31) |
| 15:30.40 | *** join/#asterisk ghento (n=ghento@bas8-toronto01-1279270360.dsl.bell.ca) |
| 15:31.18 | waKKu | folks.. there someone using pickupgroups on asterisk 1.2 with sucess? i cant get this work |
| 15:31.37 | Mercestes | robl^, Dont need the nakid men, katty doesn't appear to be online today. |
| 15:31.42 | Mercestes | so just nekkid women |
| 15:31.47 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 15:33.46 | *** join/#asterisk Fulk (n=fulk@87-194-176-39.bethere.co.uk) |
| 15:34.38 | [TK]D-Fender | [jwb]: Shouldn't |
| 15:35.20 | *** join/#asterisk blepsoaf (n=pbaker@nnat-gw.adeptra.com) |
| 15:35.28 | [jwb] | [TK]D-Fender: hrm, maybe something else is wrong with my syntax then |
| 15:35.40 | [TK]D-Fender | [jwb]: Or the file itself. |
| 15:35.44 | [jwb] | nah |
| 15:35.51 | [jwb] | because it plays fine with a Playback() before the Read() |
| 15:36.06 | [TK]D-Fender | [jwb]: Discount what you will, but you've shown us nothing :) |
| 15:36.15 | blepsoaf | hi all, is there away to allow meet me to fall through to the next in line thing in the dial plan upon entering an invalid conf #. IE http://pastebin.ca/581506 in this example is they enter an incorrect conf number it will not goto s,6 to play goodbye |
| 15:36.20 | [jwb] | I've seen it with my own eyes! ;) |
| 15:36.45 | [jwb] | I'll poke at it further, wanted to see if anyone knew of a known issue before I do |
| 15:41.02 | [TK]D-Fender | [jwb]: Yes... you have "issues", but I'm not qualified to treat you for them ;) |
| 15:41.17 | *** join/#asterisk kvit123 (n=kvit123@203.209.31.219) |
| 15:41.29 | JT | yay the Sangoma A500 is out |
| 15:41.43 | JT | now to see if they use something horrible like misdn ;) |
| 15:41.50 | CrashHD | [TK]D-Fender: he needs a gift certificate for therapy 'r us |
| 15:42.19 | [TK]D-Fender | [jwb]: exten=>s,3,Read(CONF|'adeptra/IVR-conference/enter_conf_num'|7||5) <--- NO QUOTES |
| 15:43.23 | mattfletcher | does anyone have any experience of passing a fax machine thru a tdm card? i've tried spandsp etc to no avail, and i'm hoping that i can do it this way instead |
| 15:44.29 | Fulk | mattfletcher, I've been wondering the same thing |
| 15:44.46 | *** join/#asterisk allen__s (n=chatzill@72.242.225.99) |
| 15:45.46 | [TK]D-Fender | mattfletcher: What's to know? |
| 15:46.08 | *** join/#asterisk kombi_ (n=kombi@213.160.14.18) |
| 15:46.27 | [TK]D-Fender | Sangoma A500 : whee! |
| 15:46.35 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 15:46.40 | Fulk | sorry, I've been asking about something else |
| 15:46.58 | Fulk | Fax comes in on PRI card, and I want to run a hardware fax device off an FXS |
| 15:47.10 | Fulk | that's not possible is it? |
| 15:47.21 | kombi_ | any up/downside to connecting your softphone via IAX vs. sip? |
| 15:47.21 | [TK]D-Fender | Fulk: Sure |
| 15:47.34 | Fulk | [TK]D-Fender, I thought there were timing issues in having two PCI cards? |
| 15:47.43 | *** part/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-211-202.dsl.irvnca.pacbell.net) |
| 15:47.52 | [TK]D-Fender | kombi_: fewer choices for IAX. IAX is more NAT friendly. |
| 15:47.58 | mattfletcher | [TK]D-Fender: well basically, what are the chances of it working? |
| 15:48.02 | kombi_ | thanks Fender! |
| 15:48.03 | [TK]D-Fender | Fulk: Nope. |
| 15:48.11 | coppice | if the FAX machine is on an FXS port on a channel bank attached to the same T1/E1 card as the PRI is should work. if its connected elsewhere, it will be quirky at best |
| 15:48.21 | [TK]D-Fender | Fulk: Odds depends on luck, but could be OK. |
| 15:48.30 | Fulk | coppice, that's what I thought |
| 15:48.37 | [TK]D-Fender | Fulk: For a truely business important fax, leave it the hell away from * period. |
| 15:48.38 | mattfletcher | sorry i meant a tdm pstn card, not isdn |
| 15:49.06 | [TK]D-Fender | mattfletcher: rather |
| 15:49.10 | Fulk | [TK]D-Fender, that is a shame, COTS PBX's have no problem with fax |
| 15:49.39 | coppice | something quirky in * means some people have success with spandsp, and some don't. usually its sending that fails. the same spandsp code in iaxmodem seems to keep most people happy |
| 15:50.26 | Fulk | coppice, I'm in the UK so I don't think I can use channelbanks |
| 15:51.29 | Fulk | well, E1 channel banks aren't as readily available or cheap |
| 15:51.43 | Fulk | cheaper prospect is to get another PSTN line in and have the number moved |
| 15:51.54 | coppice | in the UK? get out, man, there is still time |
| 15:52.07 | mattfletcher | so no-one here has had a positive experience with a fax machine work with a pots TDM card? |
| 15:52.24 | coppice | yeah, E1 channel banks are a very screwed up business |
| 15:52.52 | Fulk | I don't think I have any other option |
| 15:53.09 | Fulk | move line, or buy a siemens highpath PBX instead |
| 15:53.25 | blepsoaf | [TK]D-Fender: hmm i removed the quotes but it still didnt work |
| 15:53.35 | [TK]D-Fender | blepsoaf: PASTEBIN <- |
| 15:53.41 | blepsoaf | doh ok |
| 15:54.02 | Fulk | nothing wrong with the UK, better than most countries |
| 15:54.50 | coppice | I was lucky. i escaped |
| 15:54.57 | [TK]D-Fender | Fulk: Sure thing Mr. Huxley ;) |
| 15:55.33 | [TK]D-Fender | coppice: Happy Ex-Pat extraordinaire ;) |
| 15:55.50 | coppice | my name was never Pat |
| 15:56.08 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 15:56.28 | Mercestes | What's so good about the UK? |
| 15:56.50 | holiday_42 | is mr. huxley the bad guy from "elmo in grouchland"? |
| 15:57.50 | Mercestes | Japan: hot asian twins making out on camera. Denmark: no public decency laws, hot high school girls stripping on stage in school. Sweeden: blondes. Switzerland: more blondes. Russia: MailOrderBrides.com, India: really submissive chicks. |
| 15:57.53 | Mercestes | what does the UK give hte world, eh? |
| 15:58.42 | [TK]D-Fender | holiday_42: as in Aldous Huxley... |
| 15:59.01 | [TK]D-Fender | coppice: Shirley you jest! ;) |
| 15:59.36 | *** join/#asterisk matsk (i=matsk@h110n2fls32o882.telia.com) |
| 15:59.44 | [TK]D-Fender | Mercestes: So you can act decently in public in Denmark now? :) |
| 16:00.21 | blepsoaf | [TK]D-Fender: http://pastebin.ca/581544 is what i have in the DP now |
| 16:00.34 | cpm | me can't act at all |
| 16:00.40 | *** join/#asterisk ghento (n=ghento@bas8-toronto01-1279270360.dsl.bell.ca) |
| 16:00.55 | *** part/#asterisk mattfletcher (n=matt@88-97-179-134.dsl.zen.co.uk) |
| 16:01.08 | Mercestes | [TK]D-Fender, exactly! |
| 16:01.09 | rob0 | cpm: Stop acting silly. |
| 16:01.11 | [TK]D-Fender | blepsoaf: CLI <--------- |
| 16:01.23 | cpm | heh |
| 16:01.24 | blepsoaf | ok hold |
| 16:01.34 | cpm | good morn'n rob0 |
| 16:01.57 | cpm | Okay, some sell me summa dat co-lo/bandwidth, eh? rob0? got some space in yer server room? |
| 16:02.17 | rob0 | Probably could find some. |
| 16:03.14 | blepsoaf | [TK]D-Fender: http://pastebin.ca/581548 |
| 16:06.51 | Fulk | UK gave you LCD's |
| 16:06.56 | Fulk | steam power |
| 16:06.58 | Fulk | electricity generation |
| 16:06.59 | *** join/#asterisk cnet2 (n=nada@190.10.0.120) |
| 16:07.02 | Fulk | sewers |
| 16:07.09 | Fulk | metro's |
| 16:07.12 | [TK]D-Fender | blepsoaf: Ah, I see... bad conf# = DOA... see what meetmecount returns... |
| 16:07.21 | Fulk | computers |
| 16:07.26 | coppice | and don't forget Margaret Thatcher |
| 16:07.39 | cnet2 | hi guys.. I have a sip user I don't want him to be able to call forward. I tried 'cancallforward=no' but it is still forwarding. |
| 16:07.44 | Fulk | Margaret T, fixed britain |
| 16:07.57 | Fulk | quite fscked up place in the 70's :-) |
| 16:07.59 | cnet2 | .also, can I see on CLI what are the forward status? |
| 16:08.03 | Mercestes | Fulk: ...boring...boring...boring...boring...boring.... |
| 16:08.11 | Mercestes | Fulk: What about chicks? I saw nothing about chicks. |
| 16:08.14 | [TK]D-Fender | Fulk: Add Margaret Thatcher, bubonic plague, and the Spice Girls to that list... |
| 16:08.41 | Mercestes | [TK]D-Fender, and the pip |
| 16:08.42 | Fulk | nothing wrong with the plague, helps prune the the poor |
| 16:08.59 | coppice | anyone that thinks sucking all the oil out of the north sea in no time at all to prop up a bankupt economy is fixing things is kinda twisted |
| 16:09.26 | Fulk | sorted out the unions, freed the markets |
| 16:09.41 | Fulk | privatization |
| 16:09.42 | coppice | created lots of private monopolies |
| 16:10.26 | Fulk | private monoplies get created without the help of politicians |
| 16:10.28 | Fulk | look at M$ |
| 16:10.29 | Fulk | :-) |
| 16:11.18 | coppice | she oversaw a government almost as corrupt as GWB |
| 16:12.18 | Mercestes | I read the wiki on steam engine...I see nothing about England inventing anything steam engine related. =/ |
| 16:12.39 | Mercestes | "English physicist" showed up once regarding a man building a steam engine from a French physicists designs. |
| 16:12.44 | coppice | James Watt made the first practical piston steam engine |
| 16:13.11 | coppice | does Watt ring a bell? |
| 16:13.11 | Mercestes | That's hardly "bringing us" steam engines tho |
| 16:13.26 | blepsoaf | [TK]D-Fender: so I have to check the return value? |
| 16:13.42 | coppice | well, most people think he was signficant enough to name the unit of power after him |
| 16:13.53 | Mercestes | <PROTECTED> |
| 16:13.57 | tzafrir | How does the Sangoma A500 card connect to Asterisk? Through which channel driver? |
| 16:14.20 | *** join/#asterisk sci_05 (n=peter@waterfall.bestserversllc.net) |
| 16:14.42 | JT | the A500 scales up to 24 BRIs... just in case you ever want to make an ISDN phone pabx ;) |
| 16:14.45 | Mercestes | In 1769 James Watt, another member of the Lunar Society, patented the first significant improvements to the Newcomen type vacuum engine |
| 16:14.50 | Mercestes | Gainsborough believed that Watt had used his ideas for the invention, but there is no proof of this.[6] |
| 16:16.19 | Mercestes | James Watt (19 January 1736 – 19 August 1819) was a Scottish inventor |
| 16:16.38 | Fulk | JT, my old work place had an ISDN phone pabx |
| 16:16.41 | *** join/#asterisk fnordus (n=dnall@24.85.128.203) |
| 16:16.43 | Fulk | 6 ISDN2e channels |
| 16:16.48 | cpm | coppice, no but Quasimodo does |
| 16:16.58 | JT | Fulk: they're very common in Australia |
| 16:17.06 | JT | well |
| 16:17.07 | Fulk | they're common in the UK, very common |
| 16:17.15 | Fulk | ISDN2 is probably the most common small business trunk |
| 16:17.16 | JT | Fulk: were the phones proprietary? |
| 16:17.18 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-175-183.buff.east.verizon.net) |
| 16:17.19 | Mercestes | silly UK |
| 16:17.28 | JT | better than silly POTS |
| 16:17.31 | Fulk | it was an NEC switchboard |
| 16:17.36 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 16:17.45 | rob0 | Watt doesn't ring a bell! That was Pavlov!! |
| 16:17.48 | Fulk | my point was why they had 12 BRI channels, why didn't they go PRI |
| 16:17.55 | JT | Fulk: ok, well what i mean is you can use actual ISDN standard handsets, and asterisk as the PABX |
| 16:17.55 | Fulk | >8 channels it's cheaper |
| 16:18.16 | JT | Fulk: here >4 channels is cheaper if you can get Optus PRI :) |
| 16:18.29 | SuPrSluG | hello all |
| 16:18.37 | Mercestes | salute. |
| 16:18.47 | Fulk | jt, nah proprietary NEC handsets |
| 16:18.57 | Fulk | one of the reasons I like * so much is you're free to choose handsets |
| 16:19.09 | Fulk | the competition makes them so much cheaper |
| 16:19.18 | JT | Fulk: mind you, germany is one of the only places that actually uses isdn handsets |
| 16:19.34 | cpm | rob0,, no that was Quasimodo |
| 16:19.34 | Mercestes | Yea, don't be like germany |
| 16:19.39 | Fulk | NEC display handsets were £200+, $400 each |
| 16:20.03 | SuPrSluG | what can cause a delay when outbound calling on a zap channel? I'm having a weird issue w/ the callers voice not being heard for a couple of seconds when I make an outbound call |
| 16:20.30 | JT | Fulk: sounds normal |
| 16:20.37 | Mercestes | hey...what's that ...Notre Dam song? with the chorus that sounds really creepy? |
| 16:21.30 | Mercestes | that's all in a foreign language and you can't understand them anyways |
| 16:21.34 | Fulk | SuPrSluG, what hardware are you using? |
| 16:21.48 | SuPrSluG | x100p clone |
| 16:21.59 | Fulk | bin it, they're crap |
| 16:22.18 | Mercestes | SuPrSluG, using clone hardware can cause a delay when outbound dialing on a zap channel |
| 16:22.25 | kombi_ | does it make sense to have a softphone with g729? |
| 16:22.47 | Fulk | kombi_, yes - if you're connecting over a low bandwidth connection |
| 16:23.13 | Fulk | after a look of fucking about I got my x100p clone working "okayish", it's only used for emergency calling |
| 16:23.14 | Mercestes | If your connecting over a low bandwidth connection you shouldn't be using a softphone. |
| 16:23.18 | SuPrSluG | really? thanx. what would you recommend for 2-3 lines max? |
| 16:23.32 | Fulk | Mercestes, why not? |
| 16:23.33 | SuPrSluG | hardware that is |
| 16:23.44 | Fulk | I use a softphone all the time on low bandwidth |
| 16:24.10 | kombi_ | does * support g729? |
| 16:24.22 | Fulk | kombi_, yes but you have to buy a license |
| 16:24.40 | Fulk | or use the hooky free version for non-commercial use |
| 16:25.00 | Fulk | but I'd suggested buying a license off digium, since they deserve the money |
| 16:25.05 | kombi_ | guess I rather go with g711 then, less hussle.. |
| 16:25.11 | CoffeeIV | I am trying to use a voicetronix OpenLine 4 with asterix 1.2.19. Is a particular version of the voicetronix driver required ( 2.*, 3.0, 4.0) ? I am getting compile errors on chan_vpb.c |
| 16:25.22 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 16:25.22 | JT | i think most of the money goes to the patent holder, not digium |
| 16:25.30 | kombi_ | Fulk: I agree with you on the latter |
| 16:25.31 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 16:25.31 | sci_05 | kombi_ less hussle but more bandwidth |
| 16:25.39 | Fulk | kombi_, some softphones support gSM |
| 16:25.45 | Fulk | which is quite low bandwidth |
| 16:25.57 | Fulk | it's also license free |
| 16:26.01 | JT | CoffeeIV: i think you'll find almost no-one uses voicetronix stuff here |
| 16:26.05 | JT | also quite low quality |
| 16:26.06 | kombi_ | but cell quality |
| 16:26.09 | JT | g.729 sounds better |
| 16:26.31 | CoffeeIV | JT: hmmm that's too bad, I may have to email voicetronix support |
| 16:26.35 | coppice | if G.729 sounds significantly better tyou have something broken |
| 16:26.59 | JT | perhaps the version of gsm that asterisk uses is significantly broken then :) |
| 16:27.04 | Fulk | g729/gsm sound about the same to me |
| 16:27.14 | kombi_ | I'll do some tests on codecs some time, really intersting issue |
| 16:27.24 | Fulk | there's a page on voip-info.org covering them |
| 16:27.28 | Fulk | (codecs) |
| 16:27.33 | coppice | they should do. its just that GSM 06.10 takes 60% more bits to achieve that |
| 16:27.43 | kombi_ | run'em over a recording studio monitor and compare |
| 16:28.07 | JT | i'd always rather use g.711 or g.726 :) |
| 16:28.15 | *** join/#asterisk perezmeyer (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 16:28.22 | kombi_ | anyways, got to go, cheers everyone! |
| 16:28.28 | Fulk | http://www.voip-info.org/wiki/view/Bandwidth+consumption |
| 16:28.29 | coppice | those will certainly beat G.729 or GSM 06.10 |
| 16:28.50 | JT | you think? ;) |
| 16:29.37 | coppice | not everyone does. there are endless messages from G.729 fanboys saying you can't tell it from G.711 :-) |
| 16:29.50 | JT | crazy people |
| 16:29.55 | Mercestes | silly fanboys |
| 16:30.24 | coppice | I think they listened to too much heavy metal at 11 |
| 16:30.36 | Mercestes | wow, why did they wait until they were so old? |
| 16:35.00 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 16:35.20 | JT | i listened to plenty last week, beat that, Mercestes |
| 16:35.26 | *** join/#asterisk qdk (n=qdk@0x50c627be.bynxx11.adsl-dhcp.tele.dk) |
| 16:36.01 | coppice | we listen to japanese pop every day. tedious stuff for the most part |
| 16:36.10 | JT | hehe |
| 16:36.22 | JT | don't you get hk pop there? |
| 16:36.34 | Mercestes | heh, I don't listen to it often but...it makes me feel happy |
| 16:36.48 | Mercestes | I dunno if it's the upbeat music....or just knowing it's a bunch of 14 year olds... |
| 16:36.50 | Mercestes | but, meh, who cares. |
| 16:36.53 | coppice | we get lots of mandarin, korean and japanese too |
| 16:37.07 | JT | Mercestes: i was in japan last week |
| 16:38.03 | Mercestes | lucky bastard |
| 16:38.12 | coppice | the japanese singers look cute. its just a pity they don't shut up |
| 16:38.32 | JT | i can't believe that stuff, as in food/drink etc, is cheaper than australia |
| 16:39.03 | coppice | huh? food in japan is damned expensive |
| 16:39.14 | anonymouz666 | fish everyday |
| 16:39.35 | s0ck | having a blonde moment here |
| 16:39.42 | s0ck | just redone kernel and zaptel |
| 16:39.52 | s0ck | do i need to remake * too |
| 16:39.55 | s0ck | or just restart it |
| 16:40.15 | JT | coppice: it seems cheap compared to here |
| 16:40.22 | JT | well at least the convenience stores were |
| 16:40.30 | JT | seafood is much cheaper |
| 16:40.33 | coppice | where is here? |
| 16:40.34 | JT | as well |
| 16:40.36 | JT | australia |
| 16:41.06 | coppice | I never saw any cheap food or drink in japan. |
| 16:41.19 | coppice | and people complain like crazy there about its cost |
| 16:41.26 | Qwell[] | FYI, our hardware guys rock. They have this fake electronic fish tank thing... somebody recently put a webcam pointed right on it |
| 16:41.28 | coppice | especially things like fruit |
| 16:41.33 | JT | it's all relative |
| 16:41.38 | JT | yeah |
| 16:41.46 | JT | i didn't go grocery shopping |
| 16:42.45 | coppice | one of my colleagues loves fruit, and complained the only affordable kind is bananas. interestingly, those are all imports |
| 16:44.07 | Fulk | friend from Uni was from HK |
| 16:44.12 | *** join/#asterisk DirtyD (n=DigiD@ool-18bddad8.dyn.optonline.net) |
| 16:44.14 | DirtyD | Ahhhhhhh |
| 16:44.22 | DirtyD | my TDM2400p is driving me bonkers. |
| 16:44.27 | Fulk | compsci phd |
| 16:44.36 | *** join/#asterisk Waverly360 (n=Waverly3@209.12.249.243) |
| 16:44.41 | *** join/#asterisk vn (n=nostalge@bas5-quebec14-1128556688.dsl.bell.ca) |
| 16:44.41 | Fulk | had a hobby of writing songs, and one of them got quite popular back home |
| 16:44.42 | coppice | most of my friends are from HK |
| 16:44.44 | DirtyD | Fulk, you have a Phd is compsci? |
| 16:44.55 | DirtyD | s/is/in/ |
| 16:44.56 | Fulk | I think he got £100k in royalties |
| 16:45.07 | Fulk | my friend had a phd in compsci |
| 16:45.15 | Fulk | I have a BSc in compsci, haven't got round to the phd yet :-) |
| 16:45.16 | DirtyD | oh, wow.. lot's of math.. |
| 16:45.30 | JT | that's maths, you damned yanks :P |
| 16:45.37 | Fulk | predicate calculus was a bit of a pain :-) |
| 16:45.52 | JT | compsci maths is piss easy compared to end |
| 16:45.55 | JT | en |
| 16:45.59 | JT | eng :) |
| 16:46.08 | DirtyD | JT: huh? lol |
| 16:46.09 | coppice | english? |
| 16:46.12 | DirtyD | oh engineering |
| 16:46.17 | Fulk | my twin brother has a phd in theorhetical physics |
| 16:46.19 | DirtyD | coppice, I thought the same thing. |
| 16:46.35 | Fulk | http://www.ippp.dur.ac.uk/~dph3rw/thesis.pdf |
| 16:46.41 | denon | theatrical physics? |
| 16:46.44 | Fulk | I can't get past the first sentence |
| 16:46.47 | DirtyD | haha theatrical.. |
| 16:46.52 | JT | Fulk: what is this, boast about everything day? |
| 16:46.55 | DirtyD | The drama of physics |
| 16:47.05 | Fulk | huh? |
| 16:47.20 | denon | we used to make jokes about the lab guys and their theatrical phyiscs degrees |
| 16:47.22 | DirtyD | Hey, what's wrong with this TDM2400p.. it sucks! |
| 16:47.23 | *** join/#asterisk dlynes_laptop (n=dlynes@d154-20-9-152.bchsia.telus.net) |
| 16:47.23 | Fulk | can't speel :-) |
| 16:47.25 | JT | next your pet dog will have a certificate of atainment |
| 16:47.33 | Fulk | dog no, but my cat does :-) |
| 16:47.54 | DirtyD | For some reason when I bridge between zap channels the volume is reaaaaaaaaaaaally low. |
| 16:48.18 | Fulk | DirtyD, I've got no experience of TDM's - any guys here who can help? |
| 16:48.23 | Fulk | or even gals |
| 16:48.38 | DirtyD | gals are good for only one thing. |
| 16:49.03 | coppice | DirtyD: you lack imagination |
| 16:49.12 | EmleyMoor | DirtyD: Have you tried tuning the gain? |
| 16:49.42 | DirtyD | Emley: Yeah, no go.. See when I do a sip to zap it's fine.. loud even. |
| 16:50.29 | DirtyD | I think it must be some internal bug with the hardware.. |
| 16:50.40 | EmleyMoor | DirtyD: It's just Zap to Zap that's poor? |
| 16:50.44 | DirtyD | Yeah.. |
| 16:51.21 | vn | hiya, I'm looking for an analog to IAX2 adapter that I could initially use straight on the phone and eventually use it with asterisk later, any suggestion? |
| 16:51.26 | Nugget | <PROTECTED> |
| 16:51.32 | Waverly360 | Has anyone here had any experience dealing with long distance carriers requiring an access code? |
| 16:52.04 | DirtyD | Other problem I'm having is that when I Dial(Zap/1,???????) it picks up the pstn, I can hear a dialtone, but doesnt actually dial the number.. |
| 16:52.32 | DirtyD | So, the caller now has full access to an uninhibited dial-tone on my pstn.. |
| 16:52.44 | [TK]D-Fender | vn: pretty much all of the IAX2 ATA's out there are kinda cheap (crap). |
| 16:52.47 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 16:53.05 | EmleyMoor | DirtyD: Hmmm... what kind of dialing? |
| 16:53.10 | [TK]D-Fender | DirtyD: Because the number isn ANOTHER PARAMETER |
| 16:53.18 | [TK]D-Fender | isn't* |
| 16:53.19 | EmleyMoor | Ah, good point |
| 16:53.25 | EmleyMoor | Zap/1/??????? |
| 16:53.30 | [TK]D-Fender | EmleyMoor: Indeed |
| 16:53.40 | DirtyD | I just used ? in place of the number for privacy reasons. |
| 16:53.55 | EmleyMoor | Yes... but you used a , where a / was needed |
| 16:54.00 | DirtyD | ahhhhhhh. |
| 16:54.00 | [TK]D-Fender | DirtyD: Pastebin the ACTUAL line in your dialplan. |
| 16:54.38 | [TK]D-Fender | DirtyD: "," separates parameters, hence your phone number became your TIMEOUT. |
| 16:55.02 | vn | [TK]D-Fender: what would you suggest then? everybody tells me to avoir SIP cuz it's crap with NAT |
| 16:55.14 | vn | s/avoir/avoid |
| 16:55.17 | [TK]D-Fender | vn: work 99% of the time. |
| 16:55.42 | vn | what's the 1%? |
| 16:56.07 | [TK]D-Fender | vn: A few schumks with Cisco PIX's, and a tiny handful of shit D-Link routers |
| 16:56.28 | vn | aw. I got a cisco 806.. |
| 16:56.29 | [TK]D-Fender | vn: Pretty much every other el-cheapo router out there is fine. |
| 16:56.46 | [TK]D-Fender | vn: You may or may not have issues. |
| 16:56.57 | [TK]D-Fender | vn: borrow one for a test. |
| 16:57.11 | coppice | aren't pixies supposed to make trouble? :-\ |
| 16:57.16 | vn | uhm yeah |
| 16:57.29 | vn | I guess I'll get nothing by not trying |
| 16:57.37 | blepsoaf | [TK]D-Fender: will i have to wrap the meetme thing into an AGI script? or is this doable from the DP |
| 16:57.43 | [TK]D-Fender | coppice: That's why I pay "protection" money to the dryads & fae ;) |
| 16:58.05 | [TK]D-Fender | blepsoaf: maybe.... they sure didn't build that app resilient |
| 16:58.30 | blepsoaf | lol ok |
| 16:58.32 | blepsoaf | thanks |
| 16:59.17 | EmleyMoor | What's a good economical SIP/IAX2 phone with PoE capability and 3 accounts with different ringtones? |
| 16:59.51 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 17:00.16 | [TK]D-Fender | EmleyMoor: IP501 w/ PoE bundled. |
| 17:00.26 | DirtyD | when dialing using Dial() how do I do a pause... |
| 17:00.45 | [TK]D-Fender | DirtyD: Dial(zap/1/wwww12345678) |
| 17:01.00 | [TK]D-Fender | DirtyD: "w" = .5s pause |
| 17:01.03 | [TK]D-Fender | IIRC |
| 17:01.13 | vn | voip with asterisk sounds so complicated to me :\ |
| 17:01.29 | vn | like these ringtone in regexp |
| 17:01.39 | vn | s/ringtone/ringtones |
| 17:01.40 | [TK]D-Fender | vn: its not *, its CISCO's screwed up NAT implementation. |
| 17:02.16 | [TK]D-Fender | vn: What do you mean ringtones in regexp? |
| 17:02.39 | EmleyMoor | * is fun once you get used to it |
| 17:02.55 | vn | 480@-19;10(.5/.5/1) |
| 17:02.59 | vn | like that |
| 17:03.20 | [TK]D-Fender | vn: How often do you have to actually MESS with these things, and where is that specific sample from? |
| 17:04.47 | vn | from a howto |
| 17:04.58 | vn | guess its not often but still |
| 17:06.24 | s0ck | arghgh |
| 17:06.33 | s0ck | b410p wont compile now with new kernel :/ |
| 17:06.55 | tzafrir | is b410p part of latest misdn? |
| 17:07.24 | [TK]D-Fender | vn: which howto, and for what? |
| 17:07.49 | s0ck | it downloads the latest misdn when you try and make it |
| 17:09.20 | vn | [TK]D-Fender: I closed the window, don't remember sorry |
| 17:09.34 | vn | it was to set up the sound of the busy phone I think |
| 17:09.52 | [TK]D-Fender | vn: Which phone? |
| 17:10.01 | s0ck | kernel-devel is the same as extracting a kernel off kernel.org, yeh? |
| 17:10.49 | vn | [TK]D-Fender: there, look at busy tone http://extrabright.com/mywiki/Pap2Voip |
| 17:11.38 | [TK]D-Fender | vn: that has NOTHING to do with *. |
| 17:11.51 | vn | uhm..okay |
| 17:11.58 | vn | I'm noob to voip heh |
| 17:12.29 | [TK]D-Fender | vn: Thats like blaming your car stereo for shit music when its the STATION broadcasting it. |
| 17:12.43 | [TK]D-Fender | vn: Or saying your telco sucks because you bought a shitty phone. |
| 17:12.54 | vn | ;) |
| 17:13.00 | [TK]D-Fender | vn: That an ATA's config. Get your targets right! |
| 17:16.12 | vn | heh I've got like no idea where to begin |
| 17:18.02 | *** join/#asterisk _omer (n=_omer@DSL-202-59-92-141.nexlinx.net.pk) |
| 17:18.13 | *** part/#asterisk _omer (n=_omer@DSL-202-59-92-141.nexlinx.net.pk) |
| 17:19.32 | [TK]D-Fender | ~book |
| 17:19.33 | jbot | book is, like, a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 17:19.39 | [TK]D-Fender | vn: THERE |
| 17:19.45 | vn | kthx |
| 17:20.08 | [TK]D-Fender | u can has book! |
| 17:20.22 | s0ck | [TK]D-Fender: what distro you use |
| 17:20.36 | [TK]D-Fender | s0ck: CentOS Primarily, Slackware otherwise |
| 17:20.42 | s0ck | excellent |
| 17:20.44 | s0ck | kernel-devel is the same as extracting a kernel off kernel.org, yeh? |
| 17:21.00 | [TK]D-Fender | s0ck: I've never compiled my own before. |
| 17:21.02 | s0ck | thinking along the lines of 'yum install kernel-devel' |
| 17:21.36 | vn | I wonder if we can use asterisk in a VM, there's no mention of virtual machine in that book |
| 17:21.50 | s0ck | the dir format is different, i think it's why the b410p will no longer compile |
| 17:21.57 | s0ck | utter pain in the ass |
| 17:22.27 | s0ck | [TK]D-Fender: k ;/ |
| 17:22.54 | [TK]D-Fender | s0ck: You need the headers to match |
| 17:23.35 | s0ck | /usr/src/zaptel-1.2.18/misdn/drivers/isdn/hardware/mISDN/hfc_multi.c:95:26: linux/config.h: No such file or directory |
| 17:23.38 | s0ck | ^ |
| 17:24.31 | s0ck | make[2]: Entering directory `/usr/src/redhat/BUILD/kernel-2.6.21.5default <-- dunno where it's getting this dir from... |
| 17:25.13 | LOL-WUT | my asterisk just exploded |
| 17:25.46 | waKKu | maybe have some var like KERNEL_DIR to set.. or --with-include .. |
| 17:25.51 | s0ck | hmm |
| 17:25.55 | s0ck | it's getting it from $PWD |
| 17:26.09 | waKKu | weird |
| 17:26.12 | s0ck | can someone do an echo $PWD, centos preferably |
| 17:26.37 | s0ck | i think $PWD should equal the kernel devel/source dir praps |
| 17:27.06 | MindTheGap | Anyone have any clue on CDR "dst" being set to "s" when dialing from a macro? http://www.pastebin.ca/581536 has the dialplan and output... |
| 17:27.12 | waKKu | groo:/etc/asterisk# pwd |
| 17:27.12 | waKKu | /etc/asterisk |
| 17:27.13 | waKKu | groo:/etc/asterisk# echo $PWD |
| 17:27.13 | waKKu | /etc/asterisk |
| 17:27.25 | s0ck | :s |
| 17:27.44 | waKKu | i think $PWD must be pwd.. and only it |
| 17:28.12 | s0ck | dunno why the make script uses it then, it's in /usr/src/zaptel* fs |
| 17:28.20 | *** join/#asterisk javar (n=javar@69.79.134.24) |
| 17:28.24 | [TK]D-Fender | pwned |
| 17:29.24 | waKKu | s0ck there is a include/linux.h on current directory ? |
| 17:29.38 | s0ck | nein |
| 17:29.46 | waKKu | if yes.. then do PWD=$(pwd) |
| 17:29.47 | waKKu | hm.. |
| 17:29.59 | waKKu | s0ck maybe on /usr/include/linux.h ? |
| 17:30.16 | s0ck | the common kernel-devel structure houses the directory |
| 17:30.39 | waKKu | ops.. is linux/config.h |
| 17:30.50 | s0ck | cant use yum to update keren past 2.6.9 tho |
| 17:30.55 | s0ck | so i grabbed and compiled manually |
| 17:30.58 | waKKu | groo:/etc/asterisk# locate linux\/config.h |
| 17:30.59 | waKKu | /usr/include/linux/config.h |
| 17:31.13 | tzafrir | waKKu, PWD:=$(shell pwd) is for calling it from a different make |
| 17:31.28 | s0ck | one of the prerequisites of compiling zaptel seems to be 'kernel-devel' |
| 17:31.28 | tzafrir | a recusrsive make call and such |
| 17:31.45 | s0ck | which i assumed was a simple extract of the current kernel source |
| 17:31.48 | s0ck | not so, it seems |
| 17:32.07 | vn | LOL-WUT: just like...boom? |
| 17:32.18 | waKKu | to create headers u need compile it |
| 17:32.31 | LOL-WUT | just like * |
| 17:33.45 | waKKu | MindTheGap this really happen |
| 17:34.07 | waKKu | have some way to go around it.. but, of course, i dont remember |
| 17:34.12 | waKKu | i see in a book sometime.. |
| 17:34.20 | Waverly360 | [TK]D-Fender: Hey man, are you familiar with lines like this in asterisk? "PROGRESS with cause code 31 received" |
| 17:34.49 | [TK]D-Fender | Waverly360: That one's new to me. |
| 17:35.39 | Waverly360 | [TK]D-Fender: That's what I get when I try to dial a non-working through Asterisk. |
| 17:35.45 | s0ck | i have that file in the normal path waKKu |
| 17:35.50 | s0ck | not sure where it is looking for it tho |
| 17:35.59 | Waverly360 | [TK]D-Fender: My sip phone continues to ring when that happens though... |
| 17:36.28 | [jwb] | [TK]D-Fender: yep, I can't seem to get Read() to play a sound... in ael i'm suing this syntax: |
| 17:36.29 | [jwb] | <PROTECTED> |
| 17:36.35 | [jwb] | I've tried it with and without the .gsm |
| 17:36.50 | s0ck | http://pastebin.ca/581722 |
| 17:36.53 | s0ck | if it's any help |
| 17:37.45 | [TK]D-Fender | [jwb]: pastebin your exact code and CLI output (.gsm is bad, NO extensions allowed). and proof that the file is there and accessable |
| 17:38.21 | waKKu | s0ck did u compile this kernel ??? |
| 17:38.29 | s0ck | yeh |
| 17:38.55 | s0ck | please tell me you can see my glaring error |
| 17:39.13 | [jwb] | [TK]D-Fender: as I said before.. the same exact file .. ie.. Playback(goodbye); works great |
| 17:39.20 | [jwb] | so unless Read() expects its files somewhere else... |
| 17:39.26 | [TK]D-Fender | [jwb]: PASTEBIN.............. |
| 17:39.27 | waKKu | it seems like u only untar kernel source, and dont do a make menuconfig ..... |
| 17:39.35 | waKKu | to create headers .. |
| 17:39.54 | waKKu | s0ck have one file called config.h on zaptel source-dir ? |
| 17:40.12 | s0ck | nope |
| 17:40.53 | s0ck | or do you mean sling one in there |
| 17:41.06 | s0ck | haven't had to do that for any of the other kernels tho |
| 17:42.02 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:42.42 | vn | http://x100p.com/products/FXS.php <=== is that an ATA? it doesnt mention it |
| 17:44.13 | waKKu | vn hm.. native IAX .. cool ;) |
| 17:44.45 | [jwb] | [TK]D-Fender: http://pastebin.ca/581730 :P |
| 17:45.29 | waKKu | looks like an ATA for me.. maybe with a single ATA port ? (TEL / PSTN !?!?!) |
| 17:45.32 | vn | heh, I'm just looking for something in which I plug my ol' phone and in my router and voila, just need to call a voip provider...and that I can use to learn with asterisk later |
| 17:46.08 | waKKu | vn here we use linksys pap2 ... very good and price ;) |
| 17:46.20 | waKKu | but, only supports SIP |
| 17:46.45 | *** join/#asterisk jaike (n=jaike@210.5.119.2) |
| 17:47.26 | [TK]D-Fender | [jwb]: -- Executing Read("SIP/C2955-voxel-1-45ca34a8", "acode|yesterday|8|skip|3|4") in new stack |
| 17:47.35 | [TK]D-Fender | [jwb]: -rw-r--r-- 1 root root 1485 2006-01-26 13:25 /var/lib/asterisk/sounds/goodbye.gsm |
| 17:47.42 | [jwb] | oh sorry |
| 17:47.46 | [jwb] | ls'd the wrong one for ya |
| 17:47.48 | [jwb] | standby |
| 17:47.49 | [TK]D-Fender | [jwb]: Any more apples & oranges you feel like showing me? |
| 17:48.01 | waKKu | haeoihae |
| 17:48.11 | [jwb] | -rw-r--r-- 1 root root 1320 2006-01-26 13:25 /var/lib/asterisk/sounds/yesterday.gsm |
| 17:48.15 | [jwb] | better? :P |
| 17:48.17 | waKKu | [TK]D-Fender can u help me with pickupgroups hun ? :) |
| 17:48.30 | vn | waKKu: yeah but everyone tells me tu avoid SIP |
| 17:48.55 | waKKu | vn yeah.. SIP have yours problems ... |
| 17:48.58 | vn | and there [TK]D-Fender that told me my cisco 806 could be a problem |
| 17:49.01 | [TK]D-Fender | vu : convenient little french bits slipping past the rest as if typos :) |
| 17:49.18 | waKKu | but... this link u send is ONLY IAX ... check if u provider support iax users |
| 17:49.53 | vn | yup |
| 17:49.54 | waKKu | o sorry.. version 2.0 have SIP support |
| 17:50.15 | vn | I'm looking forward to go with unlimitel |
| 17:50.19 | [TK]D-Fender | vn: I don't see "skip" as an option in that command... |
| 17:50.30 | [TK]D-Fender | vn: a few of my clients use them and are very happy |
| 17:50.31 | vn | uh? |
| 17:50.41 | [TK]D-Fender | <PROTECTED> |
| 17:50.43 | vn | them being the link I pasted? |
| 17:51.11 | vn | if my understanding is good, fxs = ata? |
| 17:51.11 | waKKu | vn he did confuse ;D |
| 17:51.24 | vn | y'all lost me |
| 17:51.25 | vn | heh |
| 17:51.33 | [TK]D-Fender | option -- options are 's' , 'i', 'n' 's' to return immediately if the line is not up, 'i' to play filename as an indication tone from your indications.conf |
| 17:51.55 | waKKu | vn maybe it can help u: http://www.voip-info.org/wiki/view/FXS |
| 17:52.06 | vn | cool, a wiki |
| 17:52.31 | *** join/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 17:52.33 | [jwb] | [TK]D-Fender: show app shows: |
| 17:52.33 | [TK]D-Fender | vn: "skips" in your call to read is not legal accoring to the instructions, and the "i" in there if breoken down would seem to indicate that it will NOT treat your file as actual AUDIO |
| 17:52.33 | [jwb] | <PROTECTED> |
| 17:52.33 | [jwb] | <PROTECTED> |
| 17:52.53 | [TK]D-Fender | [jwb]: pastebin your "show application read" |
| 17:53.17 | vn | [TK]D-Fender: are you confusing me with someone else? |
| 17:53.28 | [jwb] | [TK]D-Fender: http://pastebin.ca/581742 |
| 17:53.31 | [jwb] | vn: i think he was |
| 17:53.34 | *** join/#asterisk GlobeTrotter (n=eric@ip250-10.ct.co.cr) |
| 17:53.34 | [TK]D-Fender | [jwb]: yes, I'm mixing EVERYBODY up! |
| 17:53.48 | [TK]D-Fender | Everybody : You know which comments are for whom! |
| 17:53.54 | vn | lol |
| 17:54.21 | vn | GlobeTrotter: omg, I thought I'd never see a costarican on IRC |
| 17:55.03 | GlobeTrotter | im in costa rica but not costa rican |
| 17:55.14 | vn | ok :\ |
| 17:55.16 | [TK]D-Fender | [jwb]: ok, skip that param entirely. |
| 17:55.18 | GlobeTrotter | but thanks :) |
| 17:55.21 | vn | went two times there hehe |
| 17:55.24 | [jwb] | [TK]D-Fender: standby |
| 17:56.04 | [jwb] | [TK]D-Fender: asterisk documentation ftw, as usual ;) |
| 17:56.08 | [jwb] | [TK]D-Fender: ditching that did the trick |
| 17:56.18 | *** join/#asterisk Ironhand (i=x@xyx.nl) [NETSPLIT VICTIM] |
| 17:57.43 | jaike | anyone using nagios to monitor asterisk? |
| 17:57.49 | Corydon76-home | Yep |
| 17:58.41 | jaike | corydon: which plugin you using? can you point me where you got it? |
| 17:59.06 | Corydon76-home | The generic one we got from somewhere else, and we wrote the one to monitor PRI spans |
| 17:59.31 | jaike | hmmmm, just need to check if asterisk service is running |
| 17:59.40 | AvoidingDeadlock | Corydon76-home, can you explain the unwarranted hostility? |
| 17:59.43 | jaike | the one i found was a bit old |
| 17:59.48 | bkw_ | is this much better? |
| 18:00.07 | waKKu | jaike why dont u use snmp for it ? |
| 18:00.09 | bkw_ | oh well you guys can act childish and kick/ban anyone with a differing opinion or view. |
| 18:00.10 | *** part/#asterisk bkw_ (n=brian@adsl-70-143-48-203.dsl.tul2ok.sbcglobal.net) |
| 18:00.22 | Corydon76-home | jaike: just connecting to the manager port is enough. But you need to make sure that the manager service is running |
| 18:00.42 | Corydon76-home | enable = yes in manager.conf |
| 18:01.05 | *** join/#asterisk nny (n=nny@64.203.239.83.static-pool-4.pool.hargray.net) |
| 18:01.15 | Corydon76-home | or "enabled = yes" rather |
| 18:02.06 | jaike | yeah the plugin i downloaded tries to connect to manager but cant get it to work.. thanks anyways |
| 18:02.19 | Corydon76-home | jaike: is your manager service running? |
| 18:02.27 | jaike | yep |
| 18:02.29 | Corydon76-home | jaike: that's usually what people forget |
| 18:02.54 | jaike | were using manager for a lot of things actually |
| 18:03.05 | Corydon76-home | jaike: any firewall rules? |
| 18:03.31 | Corydon76-home | check to ensure the machine running nagios can connect to that port |
| 18:03.42 | jaike | telnet to the port works fine |
| 18:03.43 | Nugget | telnet is eeeeeeevil! |
| 18:03.47 | jaike | hehe |
| 18:04.09 | Corydon76-home | jaike: and is the login correct? It's by default set as nagios/nagios |
| 18:04.12 | jaike | the plugin i got is kinda old, was hoping to find a newer version |
| 18:04.42 | jaike | yup |
| 18:04.51 | jaike | were not using default though |
| 18:04.57 | Corydon76-home | Okay, don't know what to tell you then |
| 18:05.08 | nny | quick q, i have a script that is run by asterisk to change a conf file. I have moved the system from the asterisknow stuff, to an actual Debian based install. (booo hiss :) it works great overall, but the script is not working when invoked by asterisk. I believe this is permissions issue. as it works when run as root. the error I get is Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) |
| 18:05.41 | nny | now i am running the script as an account other than asterisk, as i don't believe asterisk has shell access |
| 18:06.01 | nny | but when invoked from asterisk directly, it doesn't work as well |
| 18:06.40 | Corydon76-home | nny: check the permissions on /var/run/asterisk and /var/run/asterisk.ctl. The user you're running the script as will need to have read and execute access to the directory and rw access to the socket |
| 18:07.11 | Corydon76-home | nny: you can set those permissions in /etc/asterisk/asterisk.conf |
| 18:08.23 | *** join/#asterisk stoffell_h (n=stoffell@d51A580AB.access.telenet.be) |
| 18:08.23 | nny | ty |
| 18:08.56 | vn | something I don't get...is an FXO something you plug on an PSTN or a router/modem? |
| 18:09.05 | *** part/#asterisk nny (n=nny@64.203.239.83.static-pool-4.pool.hargray.net) |
| 18:12.27 | Corydon76-home | vn: FXO accepts dialtone. FXS generates dialtone |
| 18:12.31 | Waverly360 | Is there something significantly different about running the Dial command from and AGI script rather than running it within extensions.conf? |
| 18:12.43 | vn | Corydon76-home: yeah but on what kind of link? |
| 18:12.49 | Waverly360 | Maybe some environment variable or something that I'm not taking into consideration? |
| 18:13.03 | Corydon76-home | vn: it is its own type of link. |
| 18:13.18 | Corydon76-home | vn: commonly referred to as Loop Start |
| 18:13.57 | vn | uhm...Ju just want to search for something fxs to ethernet line (to plug in my router) |
| 18:14.07 | vn | what's the second bit commonly called? |
| 18:14.20 | Corydon76-home | Ethernet? |
| 18:14.53 | anonymouz666 | home |
| 18:14.58 | anonymouz666 | where's work? |
| 18:15.13 | vn | so I google for a fxs to ethernet adapter |
| 18:15.23 | Corydon76-home | vn: there is no such thing |
| 18:15.31 | [TK]D-Fender | vn: What is it you want to do? 1) Use an analog phone as a SIP phone, or 2) let * use your analog LINE via a SIP gateway? |
| 18:15.40 | Corydon76-home | FXS is a channelized protocol. Ethernet is packet-based |
| 18:15.56 | Corydon76-home | They are completely different |
| 18:16.03 | [TK]D-Fender | Corydon76-home: I'm quite sure he wants a boring ATA. Just playing cat&mouse with him? |
| 18:16.24 | Corydon76-home | [TK]D-Fender: no, trying to inform why it's a nonsensical request |
| 18:17.03 | Corydon76-home | vn: just get an FXO card for your Asterisk machine |
| 18:17.16 | [TK]D-Fender | Corydon76-home: Yes it may not be properly worded, but can be easily and justifiably interpreted as the net effect of an ATA with a few protocols involved but not specifically mentioned. |
| 18:17.42 | vn | wait wait...I want to beign with something plug and play and then playa sterisk later...and what I want is to use my analog phone over the IAX2 protocol |
| 18:18.03 | Corydon76-home | [TK]D-Fender: light a fire for a man, and he'll be warm for an evening; light a man on fire, and he'll be warm for the rest of his life |
| 18:18.19 | jaike | iaxy |
| 18:18.23 | [TK]D-Fender | vn: You just want an ATA then. Look at the SPA-2102. Forget IAX2 ATA's for the moment, the quiality of the ones out there currently kinda sucks |
| 18:18.36 | vn | and what about IAX? |
| 18:18.42 | *** join/#asterisk _VoiceMeUp_COM (n=_VoiceMe@modemcable159.131-56-74.mc.videotron.ca) |
| 18:18.44 | [TK]D-Fender | vn: IAX = IAX2. |
| 18:18.50 | vn | duh... |
| 18:18.56 | [TK]D-Fender | vn: the old official IAx1 is LONG dead |
| 18:19.01 | vn | amen. |
| 18:19.14 | [TK]D-Fender | vn: Whenever you see IAx mentioend, figure that its IAX2. |
| 18:19.18 | *** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir) |
| 18:19.35 | vn | dear god, I'll just give a try with the spa3k |
| 18:19.36 | mvanbaak | not on my 0.0.1-beta2 asterisk production system |
| 18:19.38 | mvanbaak | :) |
| 18:19.41 | [TK]D-Fender | vn: so seriously, just test with a SIP ATA and see if it'll do the job. Where exactly are you located? |
| 18:19.53 | vn | [TK]D-Fender: quebec city, ste-foy |
| 18:19.57 | [TK]D-Fender | vn: SPA-3K is decet, and offers 1 FXS & 1 FXO |
| 18:20.05 | Corydon76-home | mvanbaak: there are still those of us around who remember the release of 0.1.0 |
| 18:20.16 | [TK]D-Fender | vn: Ok, not QUITE next door... I'm pretty sure you've got some * nearby though. |
| 18:20.22 | mvanbaak | Corydon76-home :) |
| 18:20.24 | [TK]D-Fender | users* |
| 18:20.32 | vn | some friends do |
| 18:20.32 | waKKu | one icecream for these that help me with pickupgroups :D |
| 18:20.37 | mvanbaak | Corydon76-home: I still remember running debian with kernel 0.94 |
| 18:20.47 | vn | they're just plain confusing |
| 18:21.30 | mvanbaak | lol |
| 18:21.31 | [TK]D-Fender | mvanbaak: GOOD MORNING STARSHINE! |
| 18:21.41 | mvanbaak | [TK]D-Fender :) how are you darling ? |
| 18:21.47 | vn | well thanks for the clarifications |
| 18:21.51 | [TK]D-Fender | mvanbaak: Almost friday! |
| 18:22.01 | mvanbaak | thank god ! |
| 18:22.05 | jaike | vn: linux user? |
| 18:22.18 | vn | jaike: yeah |
| 18:22.26 | vn | www.nostalgeek.net/phpsysinfo/ |
| 18:23.12 | mvanbaak | important info: never try to use 2 sangoma A102 cards in a xen domU domain |
| 18:23.19 | mvanbaak | it's not working really ok |
| 18:23.19 | mvanbaak | ;) |
| 18:24.02 | *** join/#asterisk watchy (n=watchy@h120.184.255.206.cable.cmdn.cablelynx.com) |
| 18:25.12 | *** part/#asterisk jaike (n=jaike@210.5.119.2) |
| 18:26.40 | *** join/#asterisk rsd99 (n=chatzill@c-71-224-187-182.hsd1.pa.comcast.net) |
| 18:29.28 | *** join/#asterisk logyati (n=paulo@201.29.18.64) |
| 18:29.41 | logyati | _voicemeup_com are u there? |
| 18:30.03 | Hmmhesays | ok I can't find the voicemail map on the wiki |
| 18:31.47 | [TK]D-Fender | Hmmhesays: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+VoiceMailMain |
| 18:32.42 | watchy | anyone got the sip 2.1 for polycoms? |
| 18:34.20 | *** join/#asterisk magic_hat (n=magic_ha@h-74-2-87-16.chcgilgm.covad.net) |
| 18:34.23 | *** join/#asterisk eatmypiano (n=eatmypia@host81-155-21-242.range81-155.btcentralplus.com) |
| 18:34.50 | magic_hat | hey everyone. I need to route an inbound call from a particular number to one of my sip phones. what's the best way to accomplish that? |
| 18:35.25 | waKKu | DST/SRC,1,Dial(SIP/XXXX) |
| 18:35.25 | waKKu | maybe |
| 18:37.11 | magic_hat | waKKu: where DST is my ext and src is the caller id? |
| 18:37.43 | Jingles | from a particular #? That's fun. |
| 18:38.13 | magic_hat | Jingles, why do you say that? |
| 18:38.39 | rsd99 | i am using exten =>_1NXX-NXX-XXXX,1,Dial(SIP/(proxy)/${EXTEN}), but when i try to make a call, it says no route to host |
| 18:39.37 | Jingles | exten => _XXXX,1,Dial(SIP/####) |
| 18:39.50 | Jingles | where XXXX is the specific caller ID, and #### is the sip extension. |
| 18:40.16 | rsd99 | even for a regular phone number? |
| 18:40.21 | Jingles | yep. |
| 18:40.40 | Jingles | I've got a collection of SIP trunks coming in all with full on 10 digit phone #s. |
| 18:40.51 | Jingles | and each one gets routed to a different bit of the dialplan. |
| 18:41.57 | rsd99 | so for 10 didigs i would go exten => _XXXXXXXXXX,1,Dial(SIP/##########) |
| 18:42.14 | *** join/#asterisk ManxPower (n=manxpowe@149.sub-70-220-126.myvzw.com) |
| 18:42.22 | rsd99 | ? |
| 18:43.08 | mvanbaak | rsd99: yup |
| 18:43.22 | [TK]D-Fender | rsd99: Dashes are not allowed |
| 18:44.25 | waKKu | magic_hat: yes |
| 18:44.37 | rsd99 | do i use ${EXTEN} inplace of the #'s on the SIP/? i am new to this, and have been racking my brain. everything else i have setup and working, just making calls to 10 10 digit numbers such as my cell phone |
| 18:45.00 | mvanbaak | ${EXTEN} holds the number that matches the _XXXXXXXXX |
| 18:45.09 | [TK]D-Fender | magic_hat: Wherever your accept your incoming call, do a GotoIf on that CallerID and go do whatever you want with it. |
| 18:45.19 | mvanbaak | so if you have a number '0123456789' it will be like this: |
| 18:45.26 | *** join/#asterisk prashant_jois (n=prashant@mail.consolidated.ab.ca) |
| 18:45.29 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 18:45.36 | mvanbaak | exten => _XXXXXXXXX,1,Dial(SIP/${EXTEN}) |
| 18:45.43 | mvanbaak | call comes in |
| 18:45.47 | mvanbaak | this line matches |
| 18:45.55 | [TK]D-Fender | mvanbaak: NO |
| 18:45.56 | mvanbaak | and it will dial SIP/0123456789 |
| 18:46.02 | mvanbaak | oh dammit |
| 18:46.20 | mvanbaak | X is every digit cept 0 right |
| 18:46.21 | mvanbaak | lol |
| 18:46.26 | magic_hat | [TK]D-Fender. thanks. |
| 18:46.39 | [TK]D-Fender | mvanbaak: Getting. COLDER. |
| 18:46.53 | mvanbaak | no, 0 matches |
| 18:47.05 | [TK]D-Fender | mvanbaak: Read what he wants to do again.... |
| 18:48.05 | mvanbaak | after I grabbed another cup'o'coffee |
| 18:48.14 | blepsoaf | does anyone know for sure before I write an agi script if that meetme can handle mutiple pin attempts.. IE right now if an incorrect pin is entered meetme will hangup and not continue on with the dial plan |
| 18:49.54 | mvanbaak | back |
| 18:50.20 | prashant_jois | I have a 4 span card and I want, say, Span 1 to act as a 24 port channel bank. I configured 1-24 in zaptel.conf as being fxsks and in zapata.conf to be fxs_ks. However, when the call comes in, I'm not getting any caller ID. Is this because asterisk is not automatically parsing the caller id after the first ring? |
| 18:50.59 | [TK]D-Fender | prashant_jois: Depends on your channelbank, and the rest of your settings in zapata.conf. Pastebin it. |
| 18:51.01 | [TK]D-Fender | ~pb |
| 18:51.01 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or at paste.lisp.org |
| 18:51.04 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ |
| 18:51.06 | blepsoaf | are you setting your dchan? |
| 18:51.36 | [TK]D-Fender | blepsoaf: Not applicable, read the big print again :) |
| 18:52.44 | prashant_jois | [TK]D-Fender: there is no channel bank, I want the first span to _act_ as a 24 port channel bank. |
| 18:53.00 | *** join/#asterisk Dantix (n=Andy_CAR@host117248040.arnet.net.ar) |
| 18:53.03 | *** join/#asterisk tuxd00d (n=tuxinato@128.187.178.29) |
| 18:53.51 | magic_hat | can I send someone to a different context with gotoif, or just another priority in the current context? |
| 18:54.08 | Jingles | either is fine. |
| 18:56.46 | Dantix | hi all, I'm really new to asterisk. I've bought an openvox 1200 card, compiled its driver and modprobed it. The leds on the board are light on. But when I've conenct a phone to the FXS module I cannot ear the dialtone. Questions: is enough to load the module to obtain the dialtone at fxs connectors? or it's needed to asterisk do that work? |
| 18:56.57 | prashant_jois | [TK]D-Fender: zaptel.conf: http://pastebin.ca/581839, zapata.conf: http://pastebin.ca/581843 |
| 19:00.51 | prashant_jois | [TK]D-Fender: sorry wrong zapata.conf, here is the right one: http://pastebin.ca/581851 |
| 19:01.48 | [TK]D-Fender | prashant_jois: What signalling coming in from your telco? |
| 19:02.00 | [TK]D-Fender | magic_hat: Either |
| 19:02.58 | prashant_jois | [TK]D-Fender: signalling is actually coming from another asterisk box, but it is fxo. |
| 19:03.15 | [TK]D-Fender | prashant_jois: Ok, what are you looking to plu INTO that port? |
| 19:04.31 | DarKnesS_WolF | i can't believe it ! |
| 19:04.52 | DarKnesS_WolF | [TK]D-Fender: since this morning i can't get the option "m" for authenticate to work :-s |
| 19:04.57 | DarKnesS_WolF | it's really pain ~ |
| 19:06.35 | prashant_jois | [TK]D-Fender: I want to plug in a 24 channel T1 cable, with each channel acting as a simple analog channel. So one asterisk box is currently calling a real channel bank with 24 channels. Now I want to move that cable out of the channel bank and into another asterisk box. This new asterisk box needs to be able to handle 24 separate incoming channels as if it were a channel bank. |
| 19:08.26 | prashant_jois | [TK]D-Fender: The configuration I posted seems to work but the caller id is missing. I'm thinking that even though I configured the 24 channels as fxsks the caller id is not being automatically handled because it is not a true analog card. Is this correct? |
| 19:10.41 | [TK]D-Fender | prashant_jois: Wait.. you effectively want to link 2 * boxes together Via a T1 cable? |
| 19:10.55 | prashant_jois | [TK]D-Fender: yes exactly |
| 19:11.05 | [TK]D-Fender | prashant_jois: and you missed the line of "callerid=asreceived" |
| 19:11.11 | [TK]D-Fender | prashant_jois: Why not SIP/IAX? |
| 19:11.48 | xkev | I hav ea channel bank hanging off mine |
| 19:11.51 | xkev | ABII ftw |
| 19:12.09 | [TK]D-Fender | prashant_jois: the minimum you should use if PRI signalling betweent he two. Analog T1 is a CRAP idea. |
| 19:12.12 | xkev | ; channel bank |
| 19:12.13 | xkev | signalling=fxo_ks |
| 19:12.23 | xkev | callwaiting=no |
| 19:12.23 | xkev | context=direct_dial |
| 19:12.23 | xkev | callerid=Fax Machine <1701> |
| 19:12.23 | xkev | channel => 97 |
| 19:12.24 | xkev | etc.. |
| 19:12.36 | prashant_jois | [TK]D-Fender: because I don't have control over the configuration of the first asterisk box, and it's currently signalling to a channel bank. My preferred method of signalling would be PRI but my hands are tied. |
| 19:12.53 | prashant_jois | [TK]D-Fender: I've tried callerid=asreceived, but it doesn't seem to work |
| 19:13.12 | [TK]D-Fender | prashant_jois: An unnecessary expense in hardware and now POORLY DEPLOYED. |
| 19:13.39 | xkev | you using the bank for analog dialtone? |
| 19:13.42 | prashant_jois | [TK]D-Fender: well I can't change that |
| 19:13.49 | [TK]D-Fender | prashant_jois: Oh wait.... perhaps the other side isn't SETTING CID.... |
| 19:14.02 | [TK]D-Fender | xkev: He's not suing a channel bank. |
| 19:14.07 | *** join/#asterisk Metfan2007 (n=metfan@dsl-200-78-29-38.prod-infinitum.com.mx) |
| 19:14.25 | [TK]D-Fender | xkev: He's using a T1 CAS to link to * boxes (bleh) |
| 19:14.41 | *** join/#asterisk rantsh (n=chatzill@201.210.16.238) |
| 19:14.55 | Metfan2007 | Hi all, I have some problems trying to install the lastest asterisk-addons.1.4, can you help me? |
| 19:14.58 | prashant_jois | [TK]D-Fender: it is, because caller id is getting through to the regular channel bank |
| 19:15.01 | rantsh | Hi everybody, it's your favorite n00b once more... |
| 19:15.24 | xkev | oh wtf nm |
| 19:16.09 | rantsh | can someone tell me what does slinear mean? is that a codec or is it "raw" media??? |
| 19:16.27 | xkev | signed linear |
| 19:16.46 | mvanbaak | used by asterisk to do internal de/en-coding |
| 19:17.15 | xkev | effectively "raw" |
| 19:17.29 | xkev | got gsm, need ulaw? gsm->slin->ulaw chain |
| 19:17.34 | Metfan2007 | cp: cannot stat `.libs/libchan_h323.so.1.0.1': No such file or directory |
| 19:17.39 | Metfan2007 | any clue???? |
| 19:17.46 | rantsh | oh ok! thanks guys |
| 19:20.00 | xkev | I used some inband cas crap for linking * to legacy pbx, but setup time is ass |
| 19:20.36 | Metfan2007 | can you compile *-addons 1.4.2 ok??? |
| 19:21.11 | xkev | if two * boxes are to be stupidly trunked together, run pri with one as pri_cpe and one as pri_net |
| 19:21.51 | *** part/#asterisk Dantix (n=Andy_CAR@host117248040.arnet.net.ar) |
| 19:21.55 | prashant_jois | xkev: Ideally i would do that, but I don't have control over the first box, which is signalling cas, so the other end has to be able to accept cas |
| 19:21.58 | xkev | but I promise an ethernet switch costs less than two T400Ps |
| 19:22.19 | mvanbaak | IAX ftw |
| 19:22.34 | xkev | yeah, just get the noob running the other one to save himself hardware and do an iax trunk |
| 19:22.39 | *** join/#asterisk zeeesh (n=aadilism@202.125.143.67) |
| 19:22.42 | zeeesh | hi |
| 19:23.03 | prashant_jois | xkev: again, that side cannot be changed. |
| 19:23.16 | xkev | are you expecting cidname? |
| 19:23.19 | xkev | or just dnis/ani |
| 19:23.33 | prashant_jois | xkev: not necessarily, just number would be fine, but I'm getting nothing |
| 19:24.07 | prashant_jois | xkev: dnis would not get through anyway, I'm trapping it in extension 's' |
| 19:26.50 | xkev | tried debug channel etc? |
| 19:27.03 | xkev | erm well nm |
| 19:27.08 | xkev | that's active chans |
| 19:27.35 | xkev | all I can say is don't hit the crack pipe too much, and good luck :) |
| 19:29.21 | prashant_jois | xkev: I've never tried debug channel, let me see what happens |
| 19:30.13 | mvanbaak | brb |
| 19:30.17 | mvanbaak | Heroes on tv |
| 19:31.12 | xkev | you can debug pri spans and eval the signalling |
| 19:31.22 | xkev | inband though, not sure if zap has a knob for that |
| 19:31.25 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-41ac8225ec3de61a) |
| 19:31.25 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 19:35.39 | prashant_jois | xkev: debug channel doesn't seem to have done anything. I can't use pri debug because it is not signalling pri. |
| 19:35.49 | prashant_jois | thanks for your help anyway |
| 19:36.07 | xkev | I know cas signalling can deliver ani and bellcore caller id |
| 19:36.25 | xkev | but you have to do the ani/dnis etc with stupid digit stuff, like separate with a star |
| 19:36.39 | Cresl1n | psssh |
| 19:36.56 | xkev | is it delivering like regular pots callerid/name if you plug in a phone to the channel bank? |
| 19:36.59 | *** join/#asterisk joetester (n=joeteste@216.191.34.13) |
| 19:37.02 | Cresl1n | I've never heard of any kind of CAS protocol that will do ANI + bellcore style CID |
| 19:37.08 | Cresl1n | usually it's DTMF ANI+CID |
| 19:37.20 | key2 | Cresl1n: right |
| 19:37.35 | xkev | I run callerid over my channel banks with fxo_ks signal |
| 19:37.38 | Cresl1n | and that's your E&M and Feature Group D type trunk protocols |
| 19:38.01 | xkev | but actually, that's not doing both |
| 19:38.12 | xkev | no point in e&m for end stations |
| 19:38.36 | *** join/#asterisk clive- (n=pirch@dsl-242-140-51.telkomadsl.co.za) |
| 19:38.41 | xkev | it's been like 2 years, but yeah.. that's what I did was dtmf e&m shit from the old pbx |
| 19:40.18 | *** part/#asterisk Jingles (n=dfbarth@39.183.dowl.anc.borealisbroadband.net) |
| 19:41.32 | *** join/#asterisk dioedu (n=dioedu@201.7.117.114) |
| 19:41.35 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 19:41.35 | watchy | whats a command in * i can use to make someone enter a password before they can dialout say Long distance? |
| 19:42.54 | bkruse | agi scripts, or a DTMF read |
| 19:43.05 | bkruse | read(blah) |
| 19:43.05 | bkruse | gotoif(${blah} etc etc |
| 19:43.20 | watchy | i thought there was some type of auth command |
| 19:43.28 | dioedu | hello, i wanna use zapata.conf with realtime static configuration, but i don't know how context of channels work in this case... because with zapata.conf, the configuration is loaded in line... someone knows if is it possible ? |
| 19:43.30 | JerJer | DISA |
| 19:43.38 | clive- | watchy use agi....there is a way to do it in the dialplan, but agi is easier |
| 19:43.51 | clive- | Hi jeremy |
| 19:44.04 | DirtyD | ok |
| 19:44.07 | JerJer | but i prefer to use bkruse's method of read -> lookup in astdb (or other) -> gotoif |
| 19:44.10 | xkev | uhh |
| 19:44.17 | watchy | disa looks rather easy |
| 19:44.17 | JerJer | clive-: hi |
| 19:44.30 | bkruse | JerJer: yep, agi's for that little of a purpose could get to out of hand |
| 19:44.33 | xkev | Authenticate(<password>) |
| 19:44.35 | xkev | easy/done |
| 19:44.39 | DirtyD | when I pickup one of my FSX lines, it rings and goes immediatly to the demo.. How to I have it provide a dialtone? |
| 19:44.45 | xkev | if you just want a simple password |
| 19:44.48 | bkruse | there ya go |
| 19:45.03 | bkruse | would this be the proper place for multiple ways to skin a cat? |
| 19:45.06 | xkev | if you want user management etc, disa/agi/whatever other duct tape you have |
| 19:45.37 | watchy | xkeV: i want them to put in an ext then put in the password then drop them to a dialtone so they can dial longdistance |
| 19:45.58 | JerJer | DirtyD: set autoanswer=no something like that |
| 19:46.00 | JerJer | in zaptel |
| 19:46.03 | JerJer | er zapata |
| 19:46.07 | xkev | watchy, you want disa then |
| 19:46.32 | xkev | you can make a disa.list and set callerid, context etc per password |
| 19:46.47 | JerJer | watchy: or read, lookup pass in astdb, gotoif pass match |
| 19:47.03 | xkev | but disa makes the dialtone for you and already works :) |
| 19:47.14 | xkev | many cats, many ways |
| 19:47.28 | watchy | ah cool |
| 19:47.50 | xkev | keep in mind w/ disa (and any in-context dialing) that your dialplan should be sequence-unique so you don't have to wait $timeout etc |
| 19:48.22 | xkev | ..like you're working w/ a zap channel |
| 19:49.00 | JerJer | why have that complexity ? |
| 19:49.01 | JerJer | just straight dial the long distance number |
| 19:49.04 | JerJer | prompt for pin |
| 19:49.08 | JerJer | if match, dial |
| 19:49.20 | watchy | hrm |
| 19:49.24 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@c-67-180-78-55.hsd1.ca.comcast.net) |
| 19:49.24 | xkev | what complexity? exten => 123,1,DISA(/etc/asterisk/disa.list); ? |
| 19:49.25 | JerJer | if not match playback tt-monkies |
| 19:49.48 | watchy | wtf is tt-monkies? haha |
| 19:49.53 | JerJer | xkev: training he human to do something different - complexity |
| 19:50.18 | xkev | disa.list has '1234|dialout|"Your Mom" <NPANXX1234>' |
| 19:50.25 | *** join/#asterisk lirakis (n=etamme@65.200.191.253) |
| 19:50.35 | lirakis | hey everyone |
| 19:50.41 | xkev | ok, I see the point |
| 19:51.00 | JerJer | yeah - human dials number like they are used to |
| 19:51.01 | xkev | writing it up in dialplan allows playback of "enter password" etc, and "bzzzt" etc |
| 19:51.05 | lirakis | im trying to get a sense of the max concurrent calls people have run, and how they have achieved them.. what codec.. hardware etc. |
| 19:51.09 | JerJer | system reacts - hey wait, i want a pin |
| 19:51.13 | xkev | yah |
| 19:51.38 | xkev | disa is backdoor shit for the guy who wrote it, and those who can follow instructions ;) |
| 19:51.41 | JerJer | i haven't - used - enough dashes - in my sentences - today |
| 19:51.53 | xkev | -------------------------- here------- are------some-----more |
| 19:53.28 | xkev | the few of us who use disa (one is even sales guy) can handle dialing a DID, hearing Playback( |
| 19:53.33 | xkev | ...bleh anyway |
| 19:53.42 | MindTheGap | Anyone have any clue on CDR "dst" being set to "s" when dialing from a macro? http://www.pastebin.ca/581536 has the dialplan and output... on 1.2 it works fine, but not in 1.4... |
| 19:53.45 | xkev | but disa is bassackwards |
| 19:54.05 | JerJer | MindTheGap: yep |
| 19:54.06 | dioedu | well... hi all, how the configuration is readed in realtime static, ie, i need to put a different context by FXO channel... Is there some way to do that ? |
| 19:54.38 | MindTheGap | well JerJer, please tell me then... :) |
| 19:54.45 | JerJer | dioedu: that sucks |
| 19:55.02 | Corydon76-home | MindTheGap: but is IS the destination extension |
| 19:55.37 | dioedu | JerJer, sorry... what do you wanna say with that ? |
| 19:55.41 | watchy | pbx.c:1797 pbx_extension_helper: No application 'DigitTimeout' |
| 19:55.41 | MindTheGap | Corydon78-home, i can understand that, but how come the dst var is read only? |
| 19:56.08 | Corydon76-home | MindTheGap: Um, it always has been readonly |
| 19:56.31 | watchy | oh its deprecated |
| 19:56.32 | watchy | doh |
| 19:56.46 | Corydon76-home | It directly reflects ${EXTEN} which is "s" in a Macro |
| 19:56.57 | MindTheGap | Set(CDR(dst)="${MACRO_EXTEN:4}") was the fist thing that went trough my mind... :) |
| 19:57.29 | DarKnesS_WolF | Corydon76-home: any idea how can i use authenticate command with the "m" option? so i map a password to an accountcode? |
| 19:57.36 | lirakis | MindTheGap: yeah.. i was gonna say.. you never set ${EXTEN} |
| 19:57.39 | Corydon76-home | So in the Macro, make the first thing Goto(${MACRO_EXTEN:4},1) |
| 19:58.01 | DarKnesS_WolF | Corydon76-home: the file should be like accountcode:password ? or what .. |
| 19:58.17 | Corydon76-home | eh? |
| 19:58.20 | lirakis | so anyone have any info on how many concurrent calls they can run? |
| 19:58.33 | Corydon76-home | lirakis: yes |
| 19:58.48 | Corydon76-home | ~thebook |
| 19:58.49 | jbot | from memory, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:58.59 | Corydon76-home | Go read. |
| 19:59.23 | [TK]D-Fender | lirakis: who is "they"? |
| 19:59.34 | lirakis | Corydon76-home: i have the book thanks |
| 19:59.43 | lirakis | Corydon76-home: im looking for peoples experiences |
| 19:59.59 | lirakis | [TK]D-Fender: we are all "they" |
| 20:00.09 | DirtyD | anyone know what would cause static on the TDM2400p.. This static is to loud and it sound like a modem trying to handshake.. its bad |
| 20:00.26 | DarKnesS_WolF | Corydon76-home: the authenticate cmd there is an option "m" it's soo evil and i coun't find anyhelp with it :-s |
| 20:00.34 | [TK]D-Fender | lirakis: you didn't give any details about what gear your want us to evaluate.... |
| 20:00.40 | Corydon76-home | DarKnesS_WolF: so don't use it |
| 20:01.26 | *** join/#asterisk YonahW (n=kvirc@IGLD-83-130-71-223.inter.net.il) |
| 20:01.40 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 20:01.51 | lirakis | [TK]D-Fender: im not looking for a specific evaluation.. im looking for what your max concurrent calls are.. and what hardware you are running.. if you are transcoding etc. Im just trying to get a feel so I can do my own evaluation.. or estimation |
| 20:02.27 | *** part/#asterisk [GuS] (n=gdnet@unaffiliated/gus/x-663402) |
| 20:03.07 | lirakis | Corydon76-home: do you have a particular idea where it talks about concurrent calls in the ATFOT book? im not seeing it.. |
| 20:03.25 | Corydon76-home | lirakis: first chapter, I think |
| 20:03.44 | Corydon76-home | where it talks about hardware |
| 20:03.46 | [TK]D-Fender | lirakis: oh, US.... go check out "dimensioning" on the WIKI |
| 20:03.47 | lirakis | Corydon76-home: the table with 4 rows.. indicating info for up to 15 channels? |
| 20:04.10 | DarKnesS_WolF | Corydon76-home: :-) i have one SIP phone somewhere .. i want to assing like 6 passwords for 6 users but i don't want th passwords to be visable in the account code CDR filed do u have any other idea? |
| 20:04.19 | lirakis | Corydon76-home: ive seen people post about several hundred concurrent calls... i am interested in what they are running to achieve that kind of volume... not 15 |
| 20:04.34 | lirakis | [TK]D-Fender: ok |
| 20:04.42 | Corydon76-home | lirakis: have you run anything yet? |
| 20:05.18 | clive- | has anyone here used asterisk to talk sip to a siemens hipath? |
| 20:05.26 | lirakis | <PROTECTED> |
| 20:06.07 | lirakis | <PROTECTED> |
| 20:07.00 | *** join/#asterisk eatmypiano (n=eatmypia@host81-155-21-242.range81-155.btcentralplus.com) |
| 20:08.52 | watchy | Authenticate with a password list is very sexy |
| 20:09.00 | Corydon76-home | lirakis: the more servers, the better |
| 20:09.29 | Corydon76-home | lirakis: you could probably do that with 10, but you're probably going to need infrastructure upgrades |
| 20:10.21 | MindTheGap | Corydon, lirakis, "s,1,Goto(_0001800.|1)" sends to the right place, but now CDR shows just 0001800 not the whole number... |
| 20:10.23 | watchy | Thanks for the help with the authentication stuff guys |
| 20:10.28 | DrukenLPY | anyone know what could cause double dtmf? |
| 20:10.47 | DrukenLPY | Incorrect password '55113366' for us |
| 20:10.59 | Corydon76-home | DrukenLPY: 1.2 or 1.4? |
| 20:11.03 | *** join/#asterisk grandpapadot (n=e@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 20:11.08 | MindTheGap | although the call completes just fine |
| 20:11.19 | *** join/#asterisk _DAW_ (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 20:11.22 | DrukenLPY | 1.2.13 |
| 20:11.27 | grandpapadot | How in the @$)@$ are you guys doing International billing in asterisk? I just looked at a rate table and wow didn't know it was that confusing. |
| 20:11.28 | Corydon76-home | DrukenLPY: I'm guessing it's a Cisco on the other end? |
| 20:11.34 | DrukenLPY | aastra |
| 20:11.37 | watchy | anyone here kind enough to send me the newest poly firmware so i don't gotta call voipsupply? |
| 20:11.51 | Corydon76-home | DrukenLPY: you probably have network latency |
| 20:11.53 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-47-145.red.bezeqint.net) |
| 20:12.12 | DrukenLPY | 100ms, not too bad... |
| 20:12.41 | *** join/#asterisk exoxe (i=exoxe@ip70-171-16-60.ga.at.cox.net) |
| 20:13.01 | Corydon76-home | DrukenLPY: what are you using for DTMF? |
| 20:13.08 | Corydon76-home | RFC2833? |
| 20:13.14 | DrukenLPY | rfc2833 afaik |
| 20:13.20 | Corydon76-home | Yeah, that's the issue |
| 20:13.37 | DrukenLPY | uhmm... k, how is that the issue?? it's always worked before... |
| 20:13.45 | Corydon76-home | If packets arrive out of order, there's a potential for repeated DTMF if the packets are not reordered correctly |
| 20:14.04 | DrukenLPY | hmm... |
| 20:14.19 | Corydon76-home | It's fixed with a DTMF redesign in 1.4 |
| 20:15.09 | MindTheGap | Corydon76-home, "s,1,Goto(_0001800.|1)" sends to the right place, but now CDR shows just _0001800. not the whole number... |
| 20:15.38 | Corydon76-home | MindTheGap: you can't Goto a pattern |
| 20:18.13 | MindTheGap | ok, lets go back a little... CDR "dst" being set to "s" when dialing from a macro... you said "Goto(${MACRO_EXTEN:4},1)" but how will I know the extension if im yet to make the match? |
| 20:19.04 | *** join/#asterisk zapp-branigan (n=zapp-bra@84.79.33.1) |
| 20:19.51 | zapp-branigan | hi when i load a module codex_speex i have this problem : [codec_speex.so]Can't modify /usr/lib/asterisk/modules/codec_speex.so's text section. Use GCC option -fPIC for shared objects, please. |
| 20:20.48 | *** join/#asterisk guillote_GNU (n=guillote@host155.200-117-35.telecom.net.ar) |
| 20:25.26 | *** join/#asterisk rikstah (n=rick@rhamnett.plus.com) |
| 20:26.49 | exoxe | how would I bridge a call based on whether the called user accepts? would it be one of the options within Dial()? |
| 20:28.50 | exoxe | i.e. they'd be prompted with, To accept the call, press one, to reject the call, press 2 |
| 20:29.30 | xkev | M() |
| 20:29.31 | *** join/#asterisk anonymouz666 (n=anonymou@re366.compuland.com.br) |
| 20:29.48 | exoxe | that's what I thought |
| 20:29.49 | codefreeze | MindTheGap: what version of * are you using? |
| 20:29.57 | exoxe | oh well, I'll keep trying then |
| 20:30.03 | MindTheGap | 1.4 |
| 20:30.11 | MindTheGap | codefreeze, 1.4 |
| 20:30.47 | MindTheGap | codefreeze, can you take a look at http://www.pastebin.ca/581536 to see what im trying to do/ |
| 20:30.54 | xkev | this is called wihin a macro, hence ARG2 |
| 20:30.58 | xkev | dial,n,Dial(${chan},20,mM(findme_confirm^${ARG2}^${UNIQUEID})); |
| 20:30.59 | codefreeze | MindTheGap: OK, I just committed this morn a change to cdr_init/cdr_update that MIGHT fix your prob. Can you update and try it? |
| 20:31.09 | xkev | [macro-findme_confirm] |
| 20:31.11 | xkev | ... |
| 20:31.15 | xkev | ;exten => s,n,Read(input,xm/findme-accept,1,,2,5); |
| 20:31.18 | xkev | ;exten => s,n,GotoIf($["${input}" = "1"]?accepted,1:rejected,1); |
| 20:31.25 | xkev | exten => accepted,n,SetVar(MACRO_RESULT=); |
| 20:31.27 | xkev | exten => rejected,n,SetVar(MACRO_RESULT=CONTINUE); |
| 20:32.16 | Corydon76-home | Uh, "SetVar"? |
| 20:32.22 | xkev | this is old code |
| 20:32.27 | xkev | same diff |
| 20:32.29 | MindTheGap | codefreeze, dont know cause im using a res_ldap svn trunk based on 1.4, can I patch it? |
| 20:33.02 | xkev | I commented the called-party input stuff because it wasn't reliable |
| 20:33.11 | codefreeze | MindTheGap: I committed it to trunk also... |
| 20:33.23 | xkev | something would cause big issues when ringing certain phones/cell carriers, dtmf ignored |
| 20:33.28 | lirakis | any one used SIPp to generate test traffic? .. does it work well? |
| 20:33.29 | exoxe | xkev: thank you! |
| 20:33.32 | xkev | np |
| 20:34.53 | MindTheGap | codefreeze, the only version working for me was checked out like 1 month ago, all others are broken and main mantainer has dropped it... can you poit me the patch? |
| 20:36.24 | MindTheGap | codefreeze, although id realy like a less dramatic approach, i mean, cant anyone using 1.4 set dst on CDRs? |
| 20:36.31 | zeeesh | normally I hit my DID(access number like 207XXXXXXX) for Dial all over the world in this way. exten => _X.,1,Answer exten => _X.,2,WaitExten(15) exten => _X.,3,Dial(SIP/${EXTEN}@carrier) exten => _X.,4,Hangup …. If I want to change this way by 's' extension then how to apply what shud be the exten and what should be the second.. I tried but failed .. ? |
| 20:36.39 | MindTheGap | from within a macro... |
| 20:36.44 | MindTheGap | ? |
| 20:37.39 | codefreeze | MindTheGap: Maybe the simplest thing to do would be to just copy main/cdr.c from current version into your source (after you save away your version's cdr.c, of course)... |
| 20:38.12 | codefreeze | MindTheGap: the CDR() function won't allow you to mod dst in a CDR, just read it out. |
| 20:38.45 | codey | hmm... shouldn't i see outgoing sip calls on my asterisk console? |
| 20:38.57 | codey | xmeeting just says "user not found" |
| 20:39.01 | codey | http://slexy.org/paste/3188 |
| 20:40.46 | MindTheGap | codefreeze, ok, will try that, can you tell me what exactly this patch acomplishes? does it read dst from $MACRO_EXTEN ? |
| 20:41.27 | *** join/#asterisk Tako-san (n=Tako-san@24.108.162.254) |
| 20:41.38 | codefreeze | MindTheGap: Yes, it does (now). in update/init. It gives pref to macroexten/macrocontext, if they are set. |
| 20:45.24 | waKKu | MindTheGap from brazil, r u ? |
| 20:46.20 | *** join/#asterisk olinux (n=olinux@72.54.254.97) |
| 20:46.49 | MindTheGap | waKKu, yes I am... |
| 20:47.09 | *** join/#asterisk shido6 (i=shido6@d221-68-200.commercial.cgocable.net) |
| 20:47.15 | waKKu | MindTheGap cool.. whats region ? |
| 20:47.49 | MindTheGap | waKKu, Minas Gerais, r u brazillian? |
| 20:48.09 | waKKu | MindTheGap yeah.. from SP, but i'm living on SC (floripa) now |
| 20:48.47 | MindTheGap | codefreeze, suppose it doesnt work, how do ppl set CDRdst from a macro? |
| 20:49.45 | MindTheGap | waKKu, bacana... vc trabalha com voip? |
| 20:49.53 | waKKu | MindTheGap has a book from Flavio Gonçalves (portuguese) that explain it.. using other 2 apps that i cant remember.. i have this book in home, i'm working now |
| 20:50.16 | waKKu | MindTheGap english, or other channelll (else they will kick us ;D) |
| 20:50.21 | MindTheGap | waKKu, is there a digital version? |
| 20:50.22 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 20:50.28 | *** join/#asterisk SuperID (n=gary@c-65-96-225-97.hsd1.ma.comcast.net) |
| 20:50.50 | olinux | remote user continues to complain about sound dropping out, everyone else has excellent call quailty and no complaints, are there any tests I can run to trap the problem? |
| 20:51.05 | waKKu | i working on a big enterprise of enginering, and we use asterisk a lot here :D |
| 20:51.07 | olinux | user is on 8mbs/768kbps cable |
| 20:51.15 | waKKu | MindTheGap no.. at least i dont have |
| 20:51.31 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 20:51.40 | x86 | ok, this PAP2T-NA is about to drive me nutty |
| 20:51.47 | waKKu | lol |
| 20:52.03 | x86 | I dial **** to get into the config mode, then i hit 111# which is the option to set the IP address. |
| 20:52.09 | waKKu | calm down.. its only a small box |
| 20:52.20 | x86 | I enter the address, review it, it parrots it back to me just fine |
| 20:52.25 | MindTheGap | waKKu, would you be kind enough to scan that page? lol |
| 20:52.26 | sci_05 | olinux do they have qos setup on their network? |
| 20:52.26 | x86 | so I hit 1 to save |
| 20:52.35 | x86 | it states 'value saved' |
| 20:52.49 | x86 | then I hit 110# to check IP address value, and it says 0.0.0.0! |
| 20:53.10 | waKKu | MindTheGap haehae.. I really just need to read the name of 2 apps and tell you.. :) |
| 20:53.42 | waKKu | x86 have no reset button ? |
| 20:54.25 | MindTheGap | waKKu, do you hang around here often? maybe we could talk tomorrow, or later tonight... |
| 20:54.35 | codefreeze | MindTheGap: You might find some clever way to do it.... I'm redo-ing a number of things about CDR's in trunk. I personally think the new methods will make life easier in CDR-ville. |
| 20:54.56 | zapp-branigan | someone use gumstix ? |
| 20:54.56 | waKKu | yeah.. i bring that names tomorrow :) |
| 20:55.10 | zapp-branigan | someone use gumstix as a pbx ? |
| 20:56.18 | lirakis | ive heard of it being done zapp |
| 20:56.36 | MindTheGap | codefreeze, I see... anything I should be aware of? im using realtime ldap... |
| 20:56.50 | MindTheGap | thanks waKKu... |
| 20:57.00 | codefreeze | MindTheGap: thinking on it a little longer, it could be that the CDR is simply telling the truth. When you jump to a macro, it's always in extension 's'.... |
| 20:57.07 | Tako-san | Any thoughts on why on occasion one zap channel will get barged by another zap channel unintentionally? |
| 20:57.55 | zeeesh | guys need little help ..new to asterisk .. how to use 's' .. trying to make call through DID .. could not get success |
| 20:57.59 | zeeesh | [incoming] |
| 20:58.00 | zeeesh | exten => s,1,Answer() |
| 20:58.00 | zeeesh | exten => s,2,WaitExten(15) |
| 20:58.00 | zeeesh | exten => s,3,Dial(SIP/${EXTEN}@E1) |
| 20:58.00 | zeeesh | exten => s,4,Hangup |
| 20:58.03 | codefreeze | MindTheGap: aware of? Philosophy shift: CDR's are generated when a bridge occurs (two channels linked to pass voice). |
| 20:58.16 | zeeesh | where is the problem will anybody guide |
| 20:58.19 | codefreeze | MindTheGap: So, no conversation, no CDR. |
| 20:59.19 | MindTheGap | codefreeze yes, it is, Corydon76-home alerted me of that... thing is even if it executes a pattern match inside a macro, dst shows the _pattern, not the match... |
| 20:59.19 | waKKu | MindTheGap maybe u can help me... i'm trying to use (since morning) the pickupgroups... have no sucess... |
| 21:01.37 | waKKu | until tomorrow |
| 21:01.42 | codefreeze | MindTheGap: CDR goes away, CDR_CONTROL() will replace it.... |
| 21:02.14 | Corydon76-home | codefreeze: it does? |
| 21:02.21 | codefreeze | MindTheGap: For your dst prob, you might save the right dst val in a CDR variable.... |
| 21:02.37 | codefreeze | Corydon76-home: That's the unintended affect of the philosophy shift |
| 21:03.01 | zapp-branigan | <PROTECTED> |
| 21:03.07 | Corydon76-home | codefreeze: that we don't set variables? |
| 21:03.15 | zeeesh | by using simple calling my 1st extension like exten => _207XXX.,1,Answer .. if using 's' then where to mention my _207XX., ? will u pls |
| 21:03.40 | codefreeze | Corydon76-home: I guess I answered the wrong Question. What was the right one? |
| 21:04.00 | Corydon76-home | codefreeze: dunno |
| 21:04.34 | *** part/#asterisk dioedu (n=dioedu@201.7.117.114) |
| 21:04.43 | MindTheGap | codefreeze, yes I can, but most CDR report generation software uses dst as standard, so I would just push the problem further... i could do it in userfield but w penalties... |
| 21:04.52 | Tako-san | Anyone familiar with using channel banks? |
| 21:05.08 | MindTheGap | waKKu, i know nothing about pickupgroups, sorry... |
| 21:05.26 | anonymouz666 | anyone in here uses both a TE card with a TDM card in the same machine ? |
| 21:08.00 | codefreeze | MindTheGap: Hmmm. That's one of the probs with the current system. You just can't control the situation the way you need to. Updates to the CDR from the channel are not always under user control. This is an example. You really would like it if the CDR were NOT updated on the macro jump, others want it to be updated. I say, let the user have total control. |
| 21:09.33 | MindTheGap | codefreeze, I couldnt agree more... lets Set(CDR(everything)) !!!!! |
| 21:10.53 | olinux | sci_05 no qos it's home user with linksys wrt54g |
| 21:10.54 | codefreeze | In my new changes (branch team/murf/CDRfix5), if you want to create your own CDR, you can, and set nearly every field the way you want it, and then post it to the back end via the dialplan. |
| 21:11.00 | *** part/#asterisk Cresl1n (i=matt@nat/digium/x-41ac8225ec3de61a) |
| 21:11.15 | MindTheGap | codefreeze, things like CFIM would benefit a lot from it... |
| 21:11.51 | codefreeze | CFIM? Cross Firm Information Matching? |
| 21:12.12 | MindTheGap | lol, nope... call forwarding... |
| 21:12.41 | sci_05 | olinux I had a client that had that problem, call dropping, one way audio. I tossed a qos router at their office and all the problems where solved. |
| 21:13.24 | olinux | can you recommend a router? i dunno a good home option with that feature |
| 21:14.45 | MindTheGap | codefreeze, like if you do a call forward lookup and dial to the forwarded exten but you dont care whom exten the user picked the call from, but you want to know that his number was answered... |
| 21:14.57 | sci_05 | olinux I used a linksys RVL200, but you might want to check that router they got. You might be able to do qos setting on it (maybe with a firmware update) |
| 21:15.06 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 21:15.24 | olinux | ok thanks |
| 21:16.04 | codefreeze | MindTheGap: you totally lost me there! ;) |
| 21:16.42 | MindTheGap | codefreeze, sorry? (remember im brazillian) |
| 21:17.58 | *** part/#asterisk bapril (n=bapril@pool-70-109-158-237.cncdnh.east.verizon.net) |
| 21:19.18 | codefreeze | MindTheGap: Hey, your English is really good for an 'outlander'! I really don't have to understand all the possible situations... just want to make sure the mechanisms are general enough to handle them... |
| 21:22.19 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 21:22.42 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 21:23.37 | MindTheGap | codefreeze, the english might be, but i just understood the "philosophy shift" you mentioned 2 seconds ago.... lol... bridge done, realtime not needed, so no problems... |
| 21:25.40 | *** join/#asterisk J4k3^ (i=J4k3@138.sub-70-218-206.myvzw.com) |
| 21:25.50 | codefreeze | MindTheGap: hmmm. Haven't studied the ramifications yet for realtime... still have to look at queues and meetme's. Covered forwards pretty well... |
| 21:26.04 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 21:27.14 | MindTheGap | codefreeze, thanks for helping out... gonna try the cdr.c and maybe tomorrow i have some feedback for you... got to go now... thanks everyone... |
| 21:27.32 | codefreeze | MindTheGap: good luck! |
| 21:28.08 | MindTheGap | codefreeze, tks! |
| 21:28.46 | tuxd00d | I change my SIP port via "bingport=XXXX" and "register => NAME:PASSWORD@SERVER:PORT" but it still says port "5060" on the CLI.. What's going on? |
| 21:29.38 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 21:31.18 | olinux | nice the linksys wrt54g firmware includes QoS |
| 21:31.30 | sci_05 | olinux I thought it might |
| 21:31.30 | tuxd00d | Qwell: ping |
| 21:31.44 | *** join/#asterisk lisandropm (n=lisandro@cpe-22-76.bvconline.com.ar) |
| 21:31.47 | sci_05 | a lot of them do |
| 21:32.02 | anonymouz666 | in what condition a atxtransfer fails? |
| 21:32.10 | anonymouz666 | to use the beeper in features.conf |
| 21:32.18 | anonymouz666 | what define a failed atxtransfer? |
| 21:33.07 | *** join/#asterisk keulin (n=cray@ifth-pdcr2.infotheme.fr) |
| 21:35.07 | *** join/#asterisk _DAW_ (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 21:38.04 | *** join/#asterisk woolbeo (n=woolbeo@exchange.services.daqe.com) |
| 21:38.09 | *** join/#asterisk jm|laptop (n=jm|home@zen.jamiem.com) |
| 21:39.00 | woolbeo | I'm having some problems with a sip phone getting it's hint stuck. is there a way to reset a hint without restarting asterisk? |
| 21:39.18 | woolbeo | it's=its |
| 21:41.02 | *** join/#asterisk bkervaski (n=e@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 21:43.00 | *** join/#asterisk breanna_ (n=brea@c-71-195-248-169.hsd1.ma.comcast.net) |
| 21:43.58 | _VoiceMeUp_COM | question |
| 21:44.06 | _VoiceMeUp_COM | can 942's do SLA's ? |
| 21:47.49 | zeeesh | can i use extension 's' at the place of extension _X., ??? |
| 21:48.05 | _VoiceMeUp_COM | not same |
| 21:48.12 | _VoiceMeUp_COM | s is when not provided |
| 21:48.23 | _VoiceMeUp_COM | _X., is match any 1 + length numbers |
| 21:48.29 | *** join/#asterisk eliyahud (n=eliyahud@ool-182f9fe7.dyn.optonline.net) |
| 21:48.42 | _VoiceMeUp_COM | actualy X is one and . is one or more |
| 21:48.45 | _VoiceMeUp_COM | so i would say 2+ |
| 21:48.57 | _VoiceMeUp_COM | or maybe dot is 0+ |
| 21:49.00 | zeeesh | getting problem when using extensions _X., its working fine .. but when getting use extension 's' .. nothing evern any error msg at console? |
| 21:49.20 | _VoiceMeUp_COM | s would be not passed |
| 21:49.30 | _VoiceMeUp_COM | like Goto(BLAH) |
| 21:49.34 | *** join/#asterisk elriah (n=e@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 21:49.39 | *** part/#asterisk YonahW (n=kvirc@IGLD-83-130-71-223.inter.net.il) |
| 21:49.42 | _VoiceMeUp_COM | where X would match Goto(BLAH|123) |
| 21:49.46 | _VoiceMeUp_COM | i guess |
| 21:49.54 | eliyahud | Hi, I'm having problem where after I reboot my asterisk server, I can't dial out on an IAX channel for a few minutes.. it gives me the error NOTICE[6230]: app_dial.c:1059 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3 - No route to destination)..... anyone know why this is happening? |
| 21:50.23 | zeeesh | so dear i tried to call exten => 999,4,MeetMe(1000|M) by using _X., ,.. but failed .. somebody told me to use 's' |
| 21:50.24 | _VoiceMeUp_COM | iax show peers |
| 21:50.38 | _VoiceMeUp_COM | iax2 show registry |
| 21:50.43 | eliyahud | binfone/101132 (Unspecified) (S) 255.255.255.255 4569 Unmonitored |
| 21:51.13 | _VoiceMeUp_COM | sip debug shows waht |
| 21:51.15 | eliyahud | 66.150.120.10:4569 101132 24.47.159.231:4569 60 Registered |
| 21:51.26 | _VoiceMeUp_COM | to see the pbx to phone answers |
| 21:51.35 | _VoiceMeUp_COM | or tcpdump -s1500 -lenx |
| 21:51.37 | _VoiceMeUp_COM | port 5060 |
| 21:51.48 | eliyahud | its an iax trunk |
| 21:52.08 | _VoiceMeUp_COM | darn all these asnwers i almost felel like TK and JT |
| 21:52.12 | _VoiceMeUp_COM | yes |
| 21:52.18 | _VoiceMeUp_COM | but the phone callign is waht iax or sip ? |
| 21:52.42 | eliyahud | oh.. sometimes sip, sometimes iax |
| 21:52.47 | _VoiceMeUp_COM | the sip is easier to debug imho.. so check the PHONE --> PBX --> IAX TRUNK --> PBX -->(here) PHONE |
| 21:53.04 | _VoiceMeUp_COM | so youll see the asnwer from pbx of why its congested maybe |
| 21:53.06 | _VoiceMeUp_COM | or not |
| 21:53.32 | eliyahud | what's the sip debug command again? |
| 21:53.38 | eliyahud | was that a typo? |
| 21:53.45 | _VoiceMeUp_COM | sip debug |
| 21:53.54 | dlynes_laptop | sip debug peer peername |
| 21:53.59 | _VoiceMeUp_COM | or sip debug peer PEERNAME or sip debug ip IP |
| 21:54.09 | dlynes_laptop | Or the ip thingamajigger, too |
| 21:54.49 | sci_05 | eliyahud you might have to do a dnsmgr refresh to have the names resolve so it can start the tunnel |
| 21:54.50 | eliyahud | not sure what to look for, don't want to paste into the channel |
| 21:55.23 | dlynes_laptop | eliyahud: is the remote end for the iax2 channel behind a firewall? |
| 21:55.42 | eliyahud | ah... sci_OS is the winner |
| 21:55.48 | eliyahud | i needed to refresh dnsmgr |
| 21:55.52 | eliyahud | works now |
| 21:56.01 | dlynes_laptop | dnsmgr is something for redhat? |
| 21:56.25 | eliyahud | i have centos installed, so maybe |
| 21:56.29 | eliyahud | kind of new to asterisk |
| 21:56.33 | dlynes_laptop | ic |
| 21:56.37 | eliyahud | but its part of asterisk |
| 21:56.43 | dlynes_laptop | Oh |
| 21:56.46 | _VoiceMeUp_COM | yeah try ip instead of names |
| 21:56.53 | dlynes_laptop | Must be specific to asterisk 1.4 then |
| 21:56.56 | eliyahud | next question, is how come it doesn't refresh by itself after I reboot my computer |
| 21:56.58 | eliyahud | nope 1.2 |
| 21:57.09 | dlynes_laptop | eliyahud: there's no dnsmgr that comes with 1.2 |
| 21:57.45 | sci_05 | dlynes_laptop I thought there was but it was commented out so it wouldn't work unless you uncomment it. |
| 21:58.18 | dlynes_laptop | sci_05: Ah...I was thinking it was a separate program he was running |
| 21:58.26 | dlynes_laptop | sci_05: I guess it's just a module |
| 21:58.49 | mvanbaak | I'm off |
| 21:58.51 | sci_05 | its the dnsmgr.conf file |
| 21:58.52 | mvanbaak | latero all |
| 21:58.56 | sci_05 | later mvanbaak |
| 21:59.59 | dlynes_laptop | sci_05: ah...dns caching support in asterisk, I guess? |
| 22:00.17 | dlynes_laptop | Never actually took notice of it...I just use IPs |
| 22:00.25 | eliyahud | yeah I think its something like that |
| 22:00.40 | eliyahud | i guess if it's still a problem I'll just use IPs |
| 22:00.51 | eliyahud | don't understand why it wouldn't refresh after I rebooted though |
| 22:01.19 | eliyahud | i set the refresh interval to 30 seconds, hopefully that'll solve it permanently |
| 22:01.47 | *** part/#asterisk mog (n=mog@c-71-207-214-192.hsd1.al.comcast.net) |
| 22:02.40 | *** join/#asterisk kFuQ (n=somedude@c-67-185-123-34.hsd1.wa.comcast.net) |
| 22:03.20 | sci_05 | eliyahud it probably will, you will just see it refreshing every 30sec in the consol when your logged in |
| 22:06.52 | *** join/#asterisk BSD_Tech (n=BSDTech@adsl-69-234-211-202.dsl.irvnca.pacbell.net) |
| 22:06.57 | BSD_Tech | qusetion |
| 22:07.22 | _VoiceMeUp_COM | Asnwer |
| 22:07.31 | BSD_Tech | with asterisk and asteriskaddons being updated when is a new ver of libpri going to be released ? |
| 22:08.15 | eliyahud | thanks for the answers guys... was a big help! |
| 22:08.59 | *** join/#asterisk rmayorga_ (i=rmyorg@unaffiliated/rmayorga) |
| 22:10.21 | sci_05 | later guys |
| 22:11.46 | codey | can someone help me with this? http://slexy.org/paste/3190 |
| 22:11.58 | codey | if i have an s-extension in the sip-context |
| 22:12.02 | codey | it directly jumps to that one |
| 22:15.07 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 22:15.34 | [hC] | When using native moh (mp3), there is no way to have asterisk NOT start from the beginning of the file every time you go on hold, is there? |
| 22:16.02 | [hC] | Also, if not using native, but using mpg123 or something, whats the best way to reload when you change music? will a simple reload res_musiconhold.so do? |
| 22:16.13 | BSD_Tech | it only stats fresh in random mode I thought |
| 22:16.26 | *** join/#asterisk tsurko (n=tsurko@77.70.24.142) |
| 22:17.10 | [hC] | Well, if you think about it, its starting fresh cause theres no process constantly playing the song to attach to, it just has to start the file as you request it |
| 22:17.12 | [hC] | which makes sense. |
| 22:17.33 | _VoiceMeUp_COM | hmm think old mpg123 used to do that |
| 22:19.53 | _VoiceMeUp_COM | sound files supposed to be 8k stero or mono again ? |
| 22:20.15 | *** join/#asterisk waKKu (n=Bersirc@unaffiliated/wakku) |
| 22:20.31 | [hC] | Ive given a client access to upload moh files, which he just overwrites over the old one |
| 22:20.47 | _VoiceMeUp_COM | ok. |
| 22:20.51 | waKKu | hey MindTheGap |
| 22:20.52 | [hC] | when that happens (i presume im gonna have to use mpg123 for this, to start in the middle or whatever) but i need to reload it to pick up the new data |
| 22:21.06 | BSD_Tech | mpg123 is crap |
| 22:21.11 | BSD_Tech | old out dated |
| 22:21.13 | _VoiceMeUp_COM | yes Richard i know ;) |
| 22:21.16 | BSD_Tech | look at madplayer |
| 22:21.28 | waKKu | i'm at home.. i'll talk about macro and CDR on ur pvt... |
| 22:21.30 | _VoiceMeUp_COM | but i think it used to stream back in the days. .as it was its own spaned process |
| 22:21.40 | _VoiceMeUp_COM | but yeah its crap.. mad is the best thing i think |
| 22:21.57 | _VoiceMeUp_COM | hey Rich.. the soudnfilesare mono or stereo ? |
| 22:22.30 | BSD_Tech | I believe they are mono |
| 22:22.48 | _VoiceMeUp_COM | thanks |
| 22:22.52 | _VoiceMeUp_COM | yeah 8k |
| 22:22.57 | _VoiceMeUp_COM | 15 for stereo |
| 22:29.43 | *** join/#asterisk seele_ (n=seele@dns.datawareltda.com) |
| 22:30.01 | jamon | anyone have any ideas as to what would cause "Proxy Authentication Required" errors on incoming calls from Gizmoproject? |
| 22:30.09 | jamon | I've tried a few setups, one from gizmo's website, couple from elsewhere, same thing on all of them |
| 22:30.12 | jamon | I can make outgoing calls fine though |
| 22:30.33 | seele_ | how can I access to the voicemail from outside (IVR option)? |
| 22:30.56 | bkruse | voicemailmain() |
| 22:33.16 | codey | okay, did anyone ever have this problem? my softphone sends "h" as the extension |
| 22:33.33 | bkruse | hangup? |
| 22:34.05 | seele_ | bkruse, exten => 98,n,Goto(voicemailmain()) ?? |
| 22:34.09 | [hC] | yikes, mpg123 on one of my boxes was eating up 100% cpu! |
| 22:34.13 | [hC] | and has been for weeks |
| 22:34.14 | [hC] | heh |
| 22:35.43 | bkruse | seele_: exten = 98,n,Voicemailmain() |
| 22:36.30 | seele_ | thanks |
| 22:36.35 | seele_ | is working now |
| 22:36.59 | bkruse | ;] |
| 22:37.00 | bkruse | np |
| 22:37.23 | BSD_Tech | bbbiab |
| 22:38.07 | *** join/#asterisk codey (n=codec@p549A12A0.dip0.t-ipconnect.de) |
| 22:38.08 | codey | wtf |
| 22:38.13 | codey | asterisk just locked up my box |
| 22:38.25 | bkruse | omfg. |
| 22:38.33 | [hC] | wtflol |
| 22:38.51 | russellb | asterisk itself can not lock up your box |
| 22:39.03 | codey | actually it *did* |
| 22:39.07 | codey | asterisk is still running |
| 22:39.10 | bkruse | LIES |
| 22:39.13 | codey | but everything else is broken |
| 22:39.14 | codey | http://slexy.org/paste/3191 |
| 22:39.15 | [hC] | then the box isnt locked, heh |
| 22:39.23 | codey | after that, everything else stopped working |
| 22:39.26 | bkruse | exten => 10,1,System(init 0) ; lawl hacks |
| 22:39.31 | codey | sshd doesnt answer. |
| 22:39.46 | russellb | a userspace app can't kill the entire box |
| 22:39.53 | [hC] | how are you calling from OSS/dsp? |
| 22:39.57 | bkruse | with the exception of firefox :P |
| 22:40.50 | codey | but at least i can listen to music until the box reboots |
| 22:40.50 | dlynes_laptop | russellb: Obviously you've never seen the uniqueness of the old version of Netscrape |
| 22:41.12 | *** join/#asterisk Strom_M (n=strom@60-241-200-26.static.tpgi.com.au) |
| 22:41.19 | dlynes_laptop | russellb: Netscrape could completely lock up a Linux box so bad that you couldn't even remotely log in |
| 22:41.20 | codey | okay it's rebooting. |
| 22:41.30 | [hC] | codey: now whats it doing? |
| 22:41.33 | codey | funky. |
| 22:41.40 | codey | i dont know, hC |
| 22:41.46 | codey | seems like all phones @work are ringing |
| 22:41.59 | codey | (the number i've called was the main-queue @work) |
| 22:42.25 | [hC] | still dont know how you originated a call from the sound card. |
| 22:42.58 | codey | it's just a dial from the command line |
| 22:43.00 | Hmmhesays | dial |
| 22:43.01 | codey | that's the default |
| 22:43.03 | Strom_M | [hC]: by using the Dial command from the CLI |
| 22:43.20 | [hC] | is this a 1.4 thing? |
| 22:43.25 | Strom_M | nope |
| 22:43.30 | Strom_M | been around forever :) |
| 22:43.31 | dlynes_laptop | [hC]: no...it was there in 1.0, and 1.2 as well |
| 22:43.36 | [hC] | Connected to Asterisk 1.2.15 currently running on donkey (pid = 26903) |
| 22:43.36 | [hC] | Verbosity is at least 3 |
| 22:43.36 | [hC] | donkey*CLI> dial |
| 22:43.36 | [hC] | No such command 'dial' (type 'help' for help) |
| 22:43.52 | Strom_M | [hC]: because you don't have chan_oss or chan_alsa installed |
| 22:43.54 | [hC] | does it only show up when you load in an oss module? |
| 22:43.55 | [hC] | ahh |
| 22:43.57 | [hC] | durr. |
| 22:44.10 | [hC] | no wonder i dont know about it |
| 22:44.33 | bkruse | or console dial |
| 22:44.36 | Nugget | It still boggles my mind a bit that some people apparently have sound cards in their asterisk servers. |
| 22:44.58 | Strom_M | Nugget: it can come in handy for overhead paging |
| 22:45.02 | bkruse | Nugget: non-production |
| 22:45.35 | Strom_M | hai2u bkruse |
| 22:45.43 | [hC] | I use sound cards for overhead paging in an entire building where you need a loud speaker |
| 22:45.48 | [hC] | or a warehouse or something |
| 22:45.56 | bkruse | Strom_M: hey! |
| 22:46.01 | bkruse | hows it over on the other side of the world? |
| 22:46.23 | Strom_M | grey and wet and cold |
| 22:46.45 | bkruse | eww :/ |
| 22:46.48 | bkruse | thats not how i picture it |
| 22:46.58 | Strom_M | well it is the middle of winter over here |
| 22:47.20 | bkruse | oh |
| 22:47.21 | bkruse | right! |
| 22:47.37 | _VoiceMeUp_COM | we had 3 days of summer and its fall today |
| 22:47.40 | Nugget | umop apisdn |
| 22:47.54 | _VoiceMeUp_COM | actually summer is suppose dto be today so .. we skipped it this year |
| 22:47.54 | [hC] | whos going to astricon this year? Im going again |
| 22:48.15 | Strom_M | _VoiceMeUp_COM: you wacky canadians |
| 22:48.48 | _VoiceMeUp_COM | bah was more fun to live in mexico.. and az |
| 22:49.03 | _VoiceMeUp_COM | for the hot temps .. but women are nice here |
| 22:49.07 | seele_ | how can I block outgoing calls from a specific phones with tone command like *66 for example ?? |
| 22:49.15 | [hC] | Its nice in vancouver today, and yesterday, but this is the first so far this year. |
| 22:49.21 | _VoiceMeUp_COM | playtones ? or congestion ? |
| 22:49.46 | *** join/#asterisk vader-- (n=me@c-71-226-197-0.hsd1.nj.comcast.net) |
| 22:49.57 | codey | okay |
| 22:50.01 | codey | i'll just try it once again |
| 22:50.03 | JT | it's friggen about 3 degrees Celcius here |
| 22:50.03 | codey | to test that |
| 22:50.06 | JT | canadian weather :P |
| 22:50.29 | [hC] | its 22 here |
| 22:50.32 | [hC] | celcius |
| 22:50.40 | [hC] | JT, where are you at? |
| 22:51.03 | JT | sydney, australia |
| 22:51.16 | clive- | JT welcome to the southern hemisphere |
| 22:51.21 | codey | jip |
| 22:51.23 | codey | locked up again |
| 22:51.24 | codey | nice |
| 22:51.27 | _VoiceMeUp_COM | and global warming |
| 22:54.10 | codey | i wonder why the whole box locks up |
| 22:55.19 | dlynes_laptop | codey: might be your crappy soundcard driver, too |
| 22:55.30 | codey | well - there's no sound card at all |
| 22:55.34 | codey | maybe thats the problem :P |
| 22:55.45 | dlynes_laptop | codey: Then how do you expect to use oss? |
| 22:56.00 | codey | it's just the default-setup and i was playing around |
| 22:56.26 | codey | but there should be somthing like "Uh wait, you wanna access something that doesn't exist?" |
| 22:57.09 | dlynes_laptop | codey: You mean make it id10t proof? |
| 22:57.29 | codey | jep |
| 22:57.30 | codey | :P |
| 22:57.57 | dlynes_laptop | codey: Why not file a bug report? http://bugs.digium.com/ |
| 22:58.27 | codey | i'll do, as soon as i'm sure it is the missing soundcard ;) |
| 22:58.39 | seele_ | when I try to change my voicemail password in the phone ... the menu ask like a record menu ... and no the password change menu ... how can i solve this?? |
| 22:58.55 | codey | i should add a prompt to my box |
| 22:58.58 | codey | to reboot it via asterisk |
| 22:59.00 | flenders | JT: 3 degrees? |
| 22:59.10 | dlynes_laptop | seele_: Voicemail(), or VoicemailMain()? |
| 22:59.24 | seele_ | <PROTECTED> |
| 22:59.27 | *** join/#asterisk apardo (n=deal@49.145.217.87.dynamic.jazztel.es) |
| 22:59.45 | dlynes_laptop | seele_: make sure you have the appropriate options set in voicemail.conf to allow changing of the voicemail password |
| 23:00.28 | JT | flenders: away from the city, it was around 3 degrees this morning |
| 23:00.40 | JT | flenders: some areas were close to 0 |
| 23:00.55 | codey | dlynes_laptop: on the other hand ... this could be the problem too: /usr/share/asterisk/mohmp3/Dragostea din Tei.mp3 |
| 23:00.55 | dlynes_laptop | JT: Melbourne? |
| 23:00.56 | JT | flenders: some are still under 0, but they're more outskirts |
| 23:00.59 | JT | dlynes_laptop: sydney |
| 23:01.24 | seele_ | what option ?? |
| 23:01.28 | dlynes_laptop | codey: a filename with spaces in it? |
| 23:01.35 | codey | no, the song itself |
| 23:01.36 | codey | :P |
| 23:01.48 | dlynes_laptop | codey: I doubt it |
| 23:03.03 | *** join/#asterisk apardo (n=deal@49.145.217.87.dynamic.jazztel.es) |
| 23:03.49 | flenders | JT: last night was the first time I slept with the heater on |
| 23:04.26 | dlynes_laptop | seele_: Are you using the #include directive inside your voicemail.conf file? |
| 23:05.02 | JT | flenders: it was pretty freezing |
| 23:05.18 | seele_ | dlynes_laptop, no |
| 23:05.36 | dlynes_laptop | seele_: are any of your voicemail passwords prefaced with a '-'? |
| 23:05.59 | seele_ | dlynes_laptop, thanks |
| 23:06.58 | dlynes_laptop | seele_: ah...you had one prefaced with a '-'? |
| 23:09.01 | seele_ | no the error is the recording .... the option to change the password is 4 and the menu says 3 |
| 23:09.34 | dlynes_laptop | cool beans |
| 23:11.01 | vader-- | are there any real advantages to 1.4.x over the 1.2.x tree? |
| 23:11.20 | Waverly360 | Hey guys, is it possible to connect two asterisk boxes with a T1 crossover cable via the PRI cards? |
| 23:11.31 | Waverly360 | ...and make them talk to each other? :) |
| 23:11.47 | codey | mmh |
| 23:12.24 | JT | Waverly360: yes |
| 23:12.54 | seele_ | how can I make a phone book for incoming calls to show the Names instead of numbers?? |
| 23:13.09 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 23:13.11 | diclophis-work | hello all |
| 23:13.19 | diclophis-work | can you extend a ISDN PRI with an ethernet extender? |
| 23:13.20 | Waverly360 | JT: Can you point me to a tutorial or way to do that? |
| 23:13.37 | JT | diclophis-work: define "ethernet extender" |
| 23:13.45 | JT | Waverly360: voip-info.org |
| 23:14.03 | JT | it has info on how to make a t1 crossover cable |
| 23:14.04 | diclophis-work | like, a little plastic thing with two female rj connections |
| 23:14.14 | JT | one box will act as network, one as cpe |
| 23:14.16 | diclophis-work | that you plug the existing T1 line into |
| 23:14.27 | diclophis-work | and "extend" it with another length of wire |
| 23:14.38 | JT | diclophis-work: is this an amplifier? |
| 23:14.39 | Waverly360 | JT: I have the t1 crossover cable. Just need to know whether I'm going to have to change my wanpipe config files or if I can simply make it work by modifying zaptel and zapata |
| 23:15.05 | JT | Waverly360: the box acting as network will need to provide timing to the span instead of receive it |
| 23:15.31 | Waverly360 | JT: Is that as simple as setting the Timing number to 0 on one box, and 1 on the other? |
| 23:15.43 | JT | Waverly360: yes |
| 23:15.59 | diclophis-work | JT... no like http://service.pcconnection.com/images/inhouse/5342237_75.jpg the box on the end of that pic |
| 23:16.06 | JT | you will also need to have the correct settings in zapata.conf |
| 23:16.49 | JT | diclophis-work: might be ok depending what the box was made for, maybe not |
| 23:16.50 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 23:17.03 | diclophis-work | JT, what if its a "diy" box |
| 23:17.03 | JT | diclophis-work: i don't understand why don't simply get a longer cat5 run? |
| 23:17.20 | diclophis-work | JT, cause that would take too long for the phone company to rewire it for us |
| 23:17.30 | JT | diclophis-work: if all the pins are 1:1, sure |
| 23:17.30 | seele_ | I need to change the format of CID in the incoming calls from PSTN |
| 23:17.32 | diclophis-work | JT we are at a colo, and they run stuff from their system into our cage |
| 23:17.51 | JT | no patch panel? |
| 23:18.14 | diclophis-work | JT its all locked behind their cabinets |
| 23:18.18 | JT | only 4 pins are used, they are not the same as ethernet pins however |
| 23:18.32 | diclophis-work | which 4 pins? |
| 23:18.38 | seele_ | my actual CID format is this *LOC*4099500#->*LOC*3390000# .... and I need to show only the number, in my case 4099500 |
| 23:18.39 | Qwell[] | ~t1 |
| 23:18.40 | jbot | t1 is probably two pairs of copper wire that carry data at a rate of 1.544 Mbps. T1 lines are used to carry 24 DS-0 signals (i.e. 24 telephone conversations) or 1.536 Mbps of data. For more information see http://www.stromcarlson.com/docs/basics/t1svcfund.pdf |
| 23:18.44 | diclophis-work | should i cut the blue wire? |
| 23:18.44 | seele_ | how can I make this ??? |
| 23:18.49 | JT | make sure you extend it with twisted pair cabling, not flat cable |
| 23:19.00 | JT | diclophis-work: cut wires, why? |
| 23:19.12 | diclophis-work | JT, it was a jke |
| 23:26.14 | seele_ | bye !! |
| 23:34.14 | *** join/#asterisk adorah (n=Michael@87.69.57.73.cable.012.net.il) |
| 23:39.32 | *** join/#asterisk |dennis| (n=dennis@200.32.233.82) |
| 23:40.21 | *** join/#asterisk jaxxan (n=jaxxan@202.70.125.109) |
| 23:40.27 | jaxxan | hey guys |
| 23:40.55 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 23:41.04 | Waverly360 | JT: Ok, I have both asterisk boxes configured with a sangoma pri card in each. I have them set to E&M Wink Signalling. One is set to be the timing source, and the other to receive it. When I do a wanrouter status, however, it still shows the PRI "connecting" on each. |
| 23:41.17 | JT | err |
| 23:41.27 | JT | why on earth would you use E&M wink? |
| 23:41.42 | Waverly360 | JT: I know...it sucks..it's to replace something that's already using it. |
| 23:42.07 | Waverly360 | JT: I'm connecting an asterisk box to another PBX...but for now I'm trying to make it work with two asterisk boxes, as I don't have the other in front of me to test. |
| 23:42.08 | JT | you said you were linking 2 asterisk boxes, why can't you use whatever you want? |
| 23:42.17 | JT | i see |
| 23:42.28 | Waverly360 | JT: This is so I can learn what the heck is going on. |
| 23:43.17 | JT | well pri_cpe and pri_net should work |
| 23:43.24 | JT | dunno about linking ast with e&m |
| 23:46.00 | Waverly360 | Hmm...ok. |
| 23:46.04 | Waverly360 | I'll keep playing around with it. |
| 23:46.23 | watchy | anyone here kind enough to send me the newest poly firmware so i don't gotta call voipsupply? |
| 23:48.59 | dlynes_laptop | watchy: That's one advantage of aastra I guess...you can just download the new firmware from their website :) |
| 23:50.22 | jaxxan | so i've always used cisco phones and poe switches for my pbx implementation @ work. I find myself working on a new project that requires 150 handsets, 41 poe switches for 41 locations. this needs to be accomplished as cheaply as possible which kinda puts cisco out the door. |
| 23:50.32 | jaxxan | any suggestions on cheap handsets and poe switches ? |
| 23:51.24 | jaxxan | it's for a school network and the switches don't require vlan tagging |
| 23:51.39 | jaxxan | it'll be completely separate from their current network. |
| 23:51.41 | JT | polycoms for phones |
| 23:51.47 | JT | no evidence that ciscos are any better anyway |
| 23:52.05 | JT | switches, there's cheap stuff that does 4 PoE ports + 4 non PoE |
| 23:52.21 | jaxxan | that sounds good |
| 23:52.29 | jaxxan | let me hit up polycoms site |
| 23:52.42 | JT | jaxxan: you know cisco conference phones? |
| 23:52.54 | jaxxan | btw, i got a polycom soundstation ip 4000 which is pimp |
| 23:52.59 | JT | ah |
| 23:53.01 | jaxxan | jt: i haven't had the chance to use any |
| 23:53.10 | JT | the cisco conference phones are rebadged polycoms |
| 23:53.18 | jaxxan | lol |
| 23:53.22 | dlynes_laptop | JT: with an uglier looking case |
| 23:53.26 | JT | heh |
| 23:53.27 | CoffeeIV | I have a voicetroniz openline4 card hooked up to asterisk 1.2, and I think I have everything compiled correctly and working -- "show channeltypes" at the *CLI> lists vpb -- is there a better *CLI> command I can type to confirm I have asterisk talking to the card ? |
| 23:53.30 | jaxxan | i got mine for $640 |
| 23:53.57 | JT | jaxxan: the cheapest polycom model with PoE is about USD$105 |
| 23:54.50 | denon | you know, is it just me, or should a PoE phone in theory be cheaper? |
| 23:54.54 | denon | PoE-only, anyway |
| 23:55.03 | denon | one less DC jack and molding in the case |
| 23:55.21 | jaxxan | you'd think |
| 23:55.27 | jaxxan | i think they cost more for the convenience |
| 23:55.29 | denon | nod |
| 23:55.30 | JT | actually no, jaxxan, the cheapest is $95 |
| 23:55.37 | JT | assuming you don't need a port for the pc |
| 23:55.38 | jaxxan | which model is that jt ? |
| 23:55.49 | JT | Soundpoint IP320 |
| 23:56.01 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 23:56.01 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:57.19 | jaxxan | do you think i can use tftp for their configs ? |
| 23:57.22 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:57.36 | JT | they can use tftp, ftp, http, https, ftps for config |
| 23:58.14 | JT | tftp is the least favourable (probably because tftp sucks) |
| 23:59.22 | jaxxan | i think it's nice cause my dhcp server can specify which tftp server the phones should contact for their configs |
