00:00.04 | carl0s- | told you I was poor at Geography :) |
00:01.51 | andrejkw | So, is anyone here nice enough to help a poor guy that can't get incoming calls to work? :) |
00:03.40 | mattfletcher | carl0s: can you try ringing me again? 01524888922 |
00:03.46 | carl0s- | 1 sec |
00:04.00 | carl0s- | ringing :D |
00:04.04 | mattfletcher | yay |
00:04.16 | mattfletcher | i think i need to tell it where to go internally now |
00:04.20 | mattfletcher | brb |
00:04.21 | carl0s- | I'll hang up. save the bill :) |
00:04.33 | carl0s- | yep.. I think you will. |
00:04.54 | mattfletcher | sorry, i assumed it would be free, i will try myself again now |
00:05.04 | carl0s- | it's not a problem :D |
00:05.54 | mattfletcher | wrong context in sipgate's example i think |
00:05.59 | mattfletcher | nearly there... |
00:06.25 | carl0s- | I didn't take the extensions bit from sipgates example because I'm using trixbox.. I just took the inbound/outboud sip trunk settings from there.# |
00:06.57 | morex | G'night all |
00:08.47 | asterisk-dud | can anyone point me in the right direction for enabling call forwarding in asterisk |
00:08.51 | mattfletcher | WAHOO!!!! |
00:08.59 | asterisk-dud | like a weblink or someting |
00:09.21 | andrejkw | mattfletcher: lucky :'( |
00:09.34 | mattfletcher | sorry andrejkw |
00:09.42 | mattfletcher | thank you so much to everyone who helped me |
00:09.57 | mattfletcher | yet another reason to spread the linux love |
00:10.15 | andrejkw | aww mam :'( |
00:10.16 | mattfletcher | net job - convert the windows xp media center 2005 box to mythtv |
00:10.20 | andrejkw | my still won't ring |
00:10.40 | carl0s- | :D |
00:10.47 | carl0s- | working then? groovy :) |
00:10.55 | andrejkw | anyone willing to help me? :'( |
00:11.28 | mattfletcher | i will try andrejkw, but as you have seen i am a newbie myself |
00:11.38 | mattfletcher | who are you with? |
00:11.38 | MoutaPT | andrejkw what is your problem? |
00:11.45 | andrejkw | VoiceStick |
00:11.56 | andrejkw | I can make calls, but not recevie calls with Asterisk. |
00:11.58 | pdthome | andrejkw: just ask your question, usually if somebody can help they will |
00:12.13 | andrejkw | If I use the information directly in my PAP2, it works fine (call & receive). |
00:12.14 | mattfletcher | do they say they support asterisk? any config files examples etc? |
00:12.29 | MoutaPT | calls from FXO ? |
00:12.31 | mattfletcher | "the information"... what do they provide? |
00:12.59 | andrejkw | There is a few people online that got it to work (they proveded their configuration), but I still can't get it to work. |
00:13.11 | andrejkw | They provide the SIP proxy and the Outbound Proxy. |
00:13.18 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
00:13.20 | *** join/#asterisk AJaymn (i=AJmn@70.59.126.197) |
00:13.33 | rene- | when a channel is hangup can it be said that it was destroyed? |
00:14.04 | andrejkw | But like I said, if I use the SIP proxy and the Outbound proxy directly in my Phone Adapter it works just fine. |
00:14.30 | mattfletcher | you said you can make calls, correct? |
00:14.34 | andrejkw | Yes. |
00:14.41 | mattfletcher | what happens when you dial in? |
00:14.58 | andrejkw | I hear the ringing tone for a while and then the service voice mail comes up. |
00:15.09 | andrejkw | But while I hear ringing, the actual phone isn't. |
00:15.21 | pdthome | it could be a natting issue |
00:15.28 | mattfletcher | i found out a lot by following the debugging on the console: |
00:15.35 | mattfletcher | asterisk -rvvvvvvvv |
00:15.42 | andrejkw | I tried nat=yes, but it makes no difference. |
00:15.57 | mattfletcher | do you see anything coming in on the console? |
00:16.31 | andrejkw | Yes, when I call out stuff shows up. But when I call in, it's dead. |
00:16.58 | andrejkw | I tried doing "sip debug", and I actually see something showing up when I call in. Bunch of weird stuff. |
00:17.03 | *** join/#asterisk ivanfm_ (n=ivanfm@201.52.162.52) |
00:17.03 | pdthome | andrejkw: http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions |
00:17.15 | *** join/#asterisk postel_ (n=jp@unaffiliated/postel) |
00:17.16 | andrejkw | That says "Destroying call" all the time. |
00:17.49 | andrejkw | I have #4. |
00:17.54 | mattfletcher | that surely suggests that the calls are getting through then. can you try that command i put: asterisk -rvvvvvvvv |
00:18.13 | andrejkw | I did. |
00:18.21 | andrejkw | Still nothing, it's dead when I call in. |
00:18.28 | mattfletcher | just now, my calls were coming in, but my routing wasn't set up to make any extensions ring |
00:18.40 | pdthome | ~pb |
00:18.43 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
00:18.53 | pdthome | andrejkw: send a link to the console dump |
00:19.01 | mattfletcher | [inbound] |
00:19.01 | mattfletcher | exten => 4888922,1,Answer |
00:19.01 | mattfletcher | exten => 4888922,2,Dial(USTM/2002@2002) |
00:19.01 | *** join/#asterisk tessier_ (n=treed@gw.drjays.com) |
00:19.04 | tessier_ | Hello all! |
00:19.15 | andrejkw | pdthome: what do you mean? |
00:19.20 | mattfletcher | that is how my inbound routing works |
00:19.31 | pdthome | the wierd stuff, go to one of those links above and paste it in |
00:19.36 | andrejkw | ok |
00:19.37 | pdthome | it will give you a link you can paste in here |
00:19.56 | tessier_ | Soon I will be implementing another asterisk based phone system. I need a PRI interface and I am not going to get burned on digiums stuff again. What would most people recommend? Some level of Cisco with a PRI module? |
00:20.04 | mattfletcher | USTM as I am using a silly nortel proprietary phone i've "borrowed" from work |
00:20.52 | andrejkw | http://pastebin.ca/90741 |
00:21.03 | andrejkw | For some reason that stuff also shows up randomly. |
00:21.13 | andrejkw | Even when I don't call in, so I am not sure anymore. |
00:22.01 | pdthome | andrejkw: this is a copy of what you got when you tried calling in? |
00:22.19 | andrejkw | Yes, but even after I hang up, it just keeps showing up and flooding the console. |
00:22.30 | andrejkw | Like one every 1 - 5 minutes. |
00:23.46 | andrejkw | I am not sure if this has anything to do with the call. |
00:23.56 | andrejkw | But I am not sure, something tells me it doesn't. |
00:24.01 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
00:24.04 | mattfletcher | i'm out - means nothing to me. no point misguiding you |
00:24.38 | andrejkw | It just looks like the calls are not even reaching Asterisk. |
00:24.39 | andrejkw | At all. |
00:25.26 | andrejkw | I've been sitting in front of this all day long, and I just can;t figure it out. |
00:25.27 | harryvv | any zoneminder users here |
00:25.46 | pdthome | andrejkw: so if you turn off sip debug and make a call in you get no console messages? |
00:25.55 | andrejkw | Nope. |
00:25.58 | andrejkw | Absolutely nothing. |
00:26.28 | hads | andrejkw: Did you read the book yet? |
00:26.45 | mattfletcher | what is your external ip, have you tried port scanning yourself to see if anything can get in? |
00:26.58 | andrejkw | No, I didn't. |
00:27.08 | hads | ~thebook |
00:27.10 | jbot | [thebook] a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
00:27.10 | andrejkw | <PROTECTED> |
00:27.12 | andrejkw | oops |
00:27.15 | andrejkw | 71.57.143.216 |
00:27.18 | mattfletcher | i can for you if you have no machine outside to do it from |
00:27.28 | hads | Do yourself a favour and read the book. |
00:27.35 | andrejkw | It would be nice if you could. |
00:27.59 | andrejkw | And how is the book going to help me in this situation? |
00:28.12 | andrejkw | When I am having problems getting the SIP provider to work properly. |
00:28.22 | *** join/#asterisk P-NuT (n=P-NuT@CPE-60-227-93-75.nsw.bigpond.net.au) |
00:28.24 | hads | Erm... to understand Asterisk |
00:29.17 | MoutaPT | andrejkw: do you have an incoming context for your SIP user? |
00:29.26 | carl0s- | Well I just tried to setup my sipgate.co.uk account again, and I think I'm mostly there. The calls are being answered by Asterisk, but Asterisk is going straight to "the number you have dialled is not in service". However my only inbound route is "any CID / any DID" so I don't know why that is. |
00:29.50 | andrejkw | MoutaPT: yes |
00:30.05 | MoutaPT | sip show registry show your provider? |
00:30.26 | pdthome | carl0s-: are you sure it's going into the context you expect? |
00:30.34 | andrejkw | Nope |
00:30.42 | andrejkw | it doesn't |
00:30.46 | P-NuT | Hi all. Does anyone use cisco 7905G phones here? and has a config for one including different ringtones? |
00:31.15 | MoutaPT | andrejkw do you have the register=> user:pass@sip.provider.XXX in your sip.conf? |
00:31.42 | andrejkw | MoutaPT: nope |
00:31.49 | andrejkw | MoutaPT: Does it go under general? |
00:32.05 | MoutaPT | my * is unaccessable now, so i can't check it |
00:32.20 | MoutaPT | look over the wiki for voipbuster examples |
00:32.24 | AJaymn | Anyone use Shellshark VoIP? |
00:32.28 | MoutaPT | voipbuster is sip provider |
00:32.33 | MoutaPT | works fine with asterisk |
00:32.37 | MoutaPT | at least withe me |
00:32.51 | MoutaPT | first you should get registred |
00:33.15 | carl0s- | pdthome: I have no idea what a blooming context is. I really am going to print off that book and start reading it in bed. I'm cheating so far. Probably not even worth it. |
00:33.24 | MoutaPT | then you may think about your dialplan! I must say, as far as i know, not all cases registry is compulsory |
00:33.33 | MoutaPT | but for newbie i would recomend |
00:33.34 | andrejkw | Ok. |
00:33.52 | andrejkw | I put it in sip.conf and now it shows up. |
00:34.01 | andrejkw | But I still get no calls. |
00:34.10 | MoutaPT | http://www.voip-info.org/wiki/view/Asterisk+VoIPBuster |
00:34.24 | MoutaPT | can you outbound calls? |
00:34.24 | andrejkw | I don't have Voipbuster. |
00:34.26 | andrejkw | o.O |
00:34.33 | andrejkw | For incoming. |
00:34.37 | MoutaPT | can you outbound calls? |
00:34.42 | andrejkw | For incoming I have voipstick. |
00:34.47 | carl0s- | ah. my context should be from-trunk. I remember now. I followed a guide last time. |
00:34.49 | andrejkw | And yes, I can do outbound out of both. |
00:34.58 | andrejkw | Voipbuster and voipstick. |
00:35.04 | MoutaPT | do you have any router? |
00:35.09 | andrejkw | Yes |
00:35.15 | andrejkw | That is weird, I am getting sip_reg_timeout: |
00:35.17 | MoutaPT | does your ASterisk runs in DMZ ? |
00:35.23 | carl0s- | yup. that fixed it. |
00:35.46 | AJaymn | Whats a VoIP provider that offers CID spoofing? |
00:35.55 | MoutaPT | do you have correctly port forward 5060 for your AStBox? |
00:36.22 | andrejkw | Hmm, no I don't actually. Let me forward it. |
00:37.12 | MoutaPT | that is a MUST DO |
00:38.01 | andrejkw | Ok I did. |
00:39.22 | MoutaPT | test it now |
00:39.27 | andrejkw | nope |
00:39.29 | andrejkw | doesn't ring still |
00:39.30 | MoutaPT | look your asterisk CLI |
00:39.40 | MoutaPT | sip debug enable |
00:40.06 | andrejkw | Jan 1 04:49:31 NOTICE[3747]: chan_sip.c:5267 sip_reg_timeout: -- Registration for '1305831xxxx@i2telecom.com' timed out, trying again (Attempt #7) |
00:40.09 | *** part/#asterisk P-NuT (n=P-NuT@CPE-60-227-93-75.nsw.bigpond.net.au) |
00:40.11 | andrejkw | I keep getting that |
00:40.26 | *** join/#asterisk enjay- (i=enjay-@wsip-24-249-169-168.ph.ph.cox.net) |
00:40.37 | MoutaPT | it seems you don't have correct config in sip.conf |
00:40.53 | MoutaPT | do you have anny sipphone from this provider? |
00:41.02 | MoutaPT | just to try to register there |
00:42.16 | MoutaPT | you may need to put settings like qualify=yes |
00:42.24 | MoutaPT | insecure=very |
00:42.26 | MoutaPT | ... |
00:42.30 | *** join/#asterisk anonymouz666 (n=anonymou@20151155235.user.veloxzone.com.br) |
00:42.49 | MoutaPT | it's better to check it with your sip provider or @ wiki |
00:44.13 | MoutaPT | andrejkw did you port forward UDP on 5060 or TCP ? |
00:44.24 | andrejkw | I did both. |
00:44.27 | MoutaPT | ok |
00:44.31 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
00:45.12 | MoutaPT | you r not getting registry... until there nothing will work i think.... |
00:45.20 | riddlebox | what could I have done wrong if asterisk doesnt always give me ringback when I call someone? |
00:45.24 | MoutaPT | i must go, sorry i didn't solve your problem |
00:45.32 | andrejkw | thank you anyway |
00:48.41 | *** join/#asterisk sugardave (n=not@cpe-66-68-164-115.austin.res.rr.com) |
00:51.17 | *** join/#asterisk unit (n=doom@Toronto-HSE-ppp3780161.sympatico.ca) |
00:53.08 | *** join/#asterisk Mr-packet (n=a@222-154-239-122.adsl.xtra.co.nz) |
00:54.31 | Mr-packet | by default does asterisk bind itself to all ports on a machine when it starts, or is that configured somewhere. |
01:01.06 | Sponge_bob | anyone use a 7970 with asterisk? |
01:01.31 | Mr-packet | Sponge. not personally, but i've seen it done |
01:02.35 | Sponge_bob | Mr-packet: i"m curious to hear how it scales with asterisk |
01:02.46 | Mr-packet | your talking about a Cisco 7970? |
01:02.51 | Sponge_bob | Mr-packet: what is that H.264? somthing like that? |
01:03.02 | knarfly | I have an extension to just play moh...works great locally. but when I connect from my office it plays one songe then hangs up. conference room with moh will play all day can anyone tell me about this? |
01:04.30 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
01:04.57 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
01:05.16 | *** join/#asterisk anthonyl (n=urmom@office.midphase.com) |
01:06.27 | Sponge_bob | knarfly: what's your qualify set to? |
01:06.48 | andrejkw | How can I force the expire per SIP provider? |
01:07.49 | knarfly | yes |
01:08.24 | clyrrad | I am having a rather strange issue - for some reason when Asterisk is writing voice mail its writing to a tmp folder - what I mean is instead of writing to /var/spool/asterisk/voicemail/[context]/[extension] it is writing to /var/spool/asterisk/voicemail/[context]/[extension]/tmp - this is fine except the MWI does not work on the phone - howerver if you dial into Comedian mail it tells you that you have messages - an |
01:08.27 | knarfly | let me double check but I'm sure it's "yes" |
01:09.26 | andrejkw | How can I make the expirey different for each SIP provider? |
01:09.35 | knarfly | Sponge_bob: qualify=yes |
01:09.44 | andrejkw | All I see is defaultexpirey that goes under general, is there even a way? |
01:10.14 | clyrrad | andrejkw - you can also put that in each context which will override whats in general |
01:10.23 | andrejkw | Oh ok :D |
01:10.26 | andrejkw | Thanks then |
01:10.31 | clyrrad | NP |
01:10.44 | clyrrad | Now - Can anyone help me? :) |
01:11.58 | *** join/#asterisk eDIsonxl (n=xian-lia@mail.artdio.com.tw) |
01:12.02 | *** join/#asterisk jero (n=jero@modemcable235.87-82-70.mc.videotron.ca) |
01:12.49 | andrejkw | Hey guys, I get "Jan 1 05:22:12 NOTICE[28529]: chan_sip.c:3588 process_sdp: No compatible codec!". |
01:13.00 | Juggie | #1, fix your system date. |
01:13.01 | andrejkw | How do I figure out what codec, and how do I fix it? |
01:13.22 | Juggie | #2, make sure you allow a proper codec path between your clients |
01:13.50 | hads | #0 read the book |
01:14.07 | clyrrad | Juggie or hads - do you know the answer to my voicemail question? |
01:14.48 | Mr-packet | how can i make asterisk bind to all interfaces when it starts up |
01:14.58 | anthonyl | 0.0.0.0 |
01:15.04 | clyrrad | Mr-packet set to 0.0.0.0 |
01:15.17 | Mr-packet | where do i configure that? |
01:16.00 | hads | clyrrad: I don't know off the top of my head, what type of phone are you using? SIP/ZAP etc. |
01:16.43 | *** join/#asterisk andrejkw (n=andrejkw@c-71-57-143-216.hsd1.fl.comcast.net) |
01:16.56 | Mr-packet | clyrrad.. is that in the [general] section of manager.conf? |
01:17.02 | clyrrad | hads - the phones are all SIP phones and they work (Tested on another server) - just the MWI light is not comming on and i cant figure out why |
01:17.12 | riddlebox | does anyone have a sipura spa-2100 working with both lines? |
01:17.38 | hads | clyrrad: Do you have a mailbox=blah in sip.conf for each friend> |
01:17.58 | Sponge_bob | clyrrad: under your sip configuration does regexten match your mailbox? |
01:18.52 | clyrrad | hads - yes i have it set to [extension]@[context] |
01:19.25 | clyrrad | Sponge_bob - i do not have a regextension= parameter set - do i need that? |
01:19.40 | hads | is context other than default? |
01:19.51 | clyrrad | hads - yes |
01:19.53 | *** join/#asterisk nortex (n=barracud@adsl-69-149-173-94.dsl.amrltx.swbell.net) |
01:20.00 | clyrrad | it is NOT 'default' |
01:20.26 | hads | Have you tried, just as a test, if it will work with a mailbox in default? |
01:20.29 | clyrrad | the context is the the client id which is the phone number |
01:20.34 | Sponge_bob | clyrrad: try it. otherwise are you using macros? |
01:20.43 | clyrrad | yes I am using Macros |
01:20.59 | *** join/#asterisk xxoxx (n=xxoxx@tor/regular/xxoxx) |
01:21.07 | Sponge_bob | is it happening for all phones? |
01:21.11 | hads | "the context is the the client id which is the phone number" - you lost me. |
01:21.11 | Sponge_bob | check to see if are macro syntax is correct. |
01:21.23 | clyrrad | hads - i have not tried it under default as the macro tells it to write to the specific context |
01:21.41 | clyrrad | hads - I have the context names in voicemail.conf as the phone number |
01:21.51 | clyrrad | with the respective extensions below |
01:22.12 | hads | OK, interesting. |
01:22.15 | clyrrad | Sponge_bob - the voice mail is getting written to the right spot becase if you dial into Comedian Mail you can check and listen to the messages |
01:22.21 | clyrrad | just the MWI light is not comming on |
01:23.01 | Sponge_bob | clyrrad: correct. i had this problem before. it was my macro that was not right |
01:23.06 | Mr-packet | when i 'debug sip' to see whats happening ( why my calls are not answered ), i see my from says " From: "Jenny" <sip:6444602512@192.168.99.2>;tag=as78a1322f ". the device at the other end has no idea where 192.168.99.1 is, and is expecting the call from the address of the vtund.. how can i get asterisk to set the from, to the address of the vtund interface? |
01:23.37 | Sponge_bob | actually not the macro but the exten to call the macro |
01:23.41 | clyrrad | Sponge_bob - interesting - do you recall what you were doing wrong in your Macro? |
01:23.57 | clyrrad | and your voice mail too was able to be checkd with Comedian Mail? |
01:24.00 | clyrrad | just now MWI light? |
01:24.05 | clyrrad | no* |
01:24.20 | Sponge_bob | no light and no indication |
01:24.26 | Kryojenik | Anyone know where I can get an Israeli DID? |
01:24.41 | Sponge_bob | it was my syntax that was wrong. basically i copied another line and forgot to change some things |
01:24.43 | *** part/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
01:24.59 | Mr-packet | Kryo.. the last one got blown up yesterday |
01:25.00 | Sponge_bob | check the values you pass the marco |
01:25.09 | Kryojenik | lol... |
01:25.35 | clyrrad | Sponge_bob - ok checking |
01:26.03 | carl0s- | I just had a bad thought. |
01:26.04 | *** join/#asterisk andrejkw (n=andrejkw@c-71-57-143-216.hsd1.fl.comcast.net) |
01:26.27 | Sponge_bob | carl0s-: no i will not rub you there. :-) |
01:26.59 | carl0s- | When I get my GSM <-> SIP gateway, and set it up with a spare SIM card to have my incoming sipgate.co.uk calls forwarded over to my mobile.. well.. I'm going to lose Caller-ID functionality aren't I :( Every call will show up as being from the other mobile :( that sucks. |
01:27.03 | carl0s- | Sponge_bob: lol :) |
01:27.10 | clyrrad | exten => s,11,Voicemail(u${ARG1}@${ACCOUNTCODE}) |
01:27.14 | clyrrad | that is my syntax |
01:28.07 | Sponge_bob | hum... |
01:28.19 | Sponge_bob | clyrrad: can you paste the rest to pastebin? |
01:28.40 | clyrrad | the entire macro? |
01:29.08 | Sponge_bob | clyrrad: does the problem exist for all sip phones? |
01:29.22 | clyrrad | yes on all phones |
01:29.38 | Sponge_bob | have you tried the regexten? |
01:29.44 | clyrrad | nope |
01:29.49 | clyrrad | how would i set that? |
01:29.50 | Sponge_bob | try it first |
01:29.56 | clyrrad | to just the extension? |
01:30.00 | clyrrad | or extension@context? |
01:30.21 | Sponge_bob | *i think* regexten=what ever exten the sip is |
01:30.35 | Sponge_bob | that goes under the sip.conf |
01:30.43 | clyrrad | ok i am adding that now |
01:31.10 | Sponge_bob | so if your extension is 100 you put regexten=100 |
01:31.22 | *** join/#asterisk overworked554 (n=overwork@209.242.52.25) |
01:32.34 | clyrrad | ok i have added that - and still no change |
01:32.47 | Sponge_bob | realod? |
01:32.50 | clyrrad | yes |
01:33.02 | Sponge_bob | pm me your macro |
01:33.07 | clyrrad | ok |
01:35.30 | *** join/#asterisk NoRemorse (n=bah@eth2462.vic.adsl.internode.on.net) |
01:35.33 | NoRemorse | hi all |
01:35.56 | NoRemorse | When I call someone on a sip client with call waiting enabled, I dont get a ring time just silence, any idea how this can be fixed please? |
01:36.06 | NoRemorse | *ring tone |
01:38.58 | SplasPood | hah thats pretty slick.. *67 is useless |
01:39.54 | andrejkw | Hey guys |
01:41.26 | NoRemorse | When I call someone on a sip client with call waiting enabled, I dont get a ring tone just silence, any idea how this can be fixed please? |
01:45.32 | *** join/#asterisk trbldwine (n=trbldwin@c-71-194-161-170.hsd1.il.comcast.net) |
01:47.33 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-42-234.losaca.adelphia.net) |
01:53.25 | NoRemorse | When I call someone on a sip client with call waiting enabled, I dont get a ring tone just silence, any idea how this can be fixed please? |
01:54.06 | NoRemorse | or is it a function of the client hardware |
01:58.46 | *** join/#asterisk babyju (n=babyju@h-67-102-255-186.nycmny83.covad.net) |
02:00.45 | clyrrad | I am having a rather strange issue - for some reason when a person leaves a voice mail the MWI is not activated - Asterisk records the voice mail and if you dial into Comedian Mail you can check the message - just the MWI light is not comming on - does anyone know what could be the cause of this? |
02:03.25 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
02:04.24 | *** join/#asterisk file2 (n=IrcNet@out.clearnet.com) |
02:04.24 | *** mode/#asterisk [+o file2] by ChanServ |
02:04.33 | file2 | mooo |
02:04.50 | Qwell | lame |
02:09.42 | andrejkw | Hey guys |
02:09.58 | andrejkw | I have a small problem, when I hang up, it doesn't get hung up on the other side. |
02:10.16 | enjay- | speakerphone? |
02:10.48 | clyrrad | Qwell - do you have any ideas for me? |
02:10.53 | enjay- | you hear the beep/beep/beep/beep when one side hangs up is that what you are referring to? |
02:11.05 | clyrrad | andrewjwk - most likely a Firewall issue |
02:11.15 | andrejkw | Yes |
02:11.24 | enjay- | Yes to my question or what? |
02:11.30 | andrejkw | exactly, if the toher side stays on long enought he beep beep beep noise comes up |
02:11.48 | enjay- | thats because it HAS been hung up and you are still on speakerphone with no peer.. |
02:11.58 | andrejkw | no speakerphone |
02:12.03 | enjay- | well handset on then.. |
02:12.10 | andrejkw | no, cellphone |
02:12.11 | andrejkw | :\ |
02:12.18 | enjay- | ah.. |
02:12.23 | enjay- | yea cell should automatically hang up.. |
02:12.31 | andrejkw | well it doesn't, it sites there |
02:12.33 | andrejkw | *sits |
02:12.43 | andrejkw | forever... |
02:12.52 | andrejkw | waiting for the other side to respond. |
02:13.08 | enjay- | hmm |
02:13.29 | andrejkw | and then I can't call the other side again' |
02:13.35 | AJaymn | Anyone use ShellShark for a Provider? |
02:13.35 | andrejkw | until I restart Asterisk |
02:14.25 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-59-1.cybersurf.com) |
02:14.35 | enjay- | you have a hangup macro defined? |
02:14.38 | enjay- | or statement rather.. |
02:15.10 | andrejkw | Yes |
02:15.43 | enjay- | might want to pastebin your output of the entire call process.. |
02:15.44 | NoRemorse | hey guys I am trying to upgrade asterisk from 1.0.9 to 1.2.10. I have taken the "delete the old evrsion completely " approach, installed zaptel 1.2.7 and libpri 1.2.3 beforehand and addons 1.2.3 after, and everything runs fine until.... I create a default cdr_mysql.conf and I get a seg fault. I have the same install on another box and it works fine, what could be polluting my new install? |
02:16.09 | NoRemorse | the old 1.0.9 talked to a local mysql 4 database |
02:17.24 | andrejkw | it never hangs up |
02:18.25 | *** join/#asterisk kuku5 (n=kuku5@c-71-201-213-102.hsd1.il.comcast.net) |
02:18.38 | kuku5 | I cant seem to stay registered with broadvoice - any suggestions? |
02:19.08 | Qwell | switch providers |
02:23.58 | *** join/#asterisk zepmantra (i=what@125.212.110.117) |
02:24.26 | kuku5 | Got SIP response 409 "Conflict" back from 147.135.12.128 |
02:24.32 | [andromeda] | Does anyone know a VoIP service that offers Toll-Free numbers, and will work with asterisk? |
02:24.46 | kuku5 | and NOTICE[21659]: chan_sip.c:4045 sip_reg_timeout: |
02:25.57 | [andromeda] | NoRemorse: the addons1.2.3 package does not compile the mysql function for asterisk correctly, it's missing some headers |
02:28.08 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
02:30.20 | *** join/#asterisk P-NuT (n=P-NuT@CPE-60-227-93-75.nsw.bigpond.net.au) |
02:31.46 | *** join/#asterisk c1sco (n=loke@c-24-5-215-69.hsd1.ca.comcast.net) |
02:32.08 | c1sco | hey guys im having trouble making inbound calls via the pstn to my asterisk box |
02:32.28 | c1sco | the asterisk box receives the call, but the phone on the pstn never gets a response from the asterisk box |
02:32.56 | c1sco | i believe i have something misconfigure in extensions.conf |
02:35.25 | clyrrad | what does the CLI say when it receives the call? |
02:35.37 | c1sco | should i paste it here? |
02:35.48 | clyrrad | if its not more than 2 lines |
02:35.52 | clyrrad | otherwise pastebin it |
02:36.21 | c1sco | <PROTECTED> |
02:36.21 | c1sco | <PROTECTED> |
02:36.21 | c1sco | <PROTECTED> |
02:36.34 | enjay- | is the phone not ringing? |
02:36.36 | c1sco | yes |
02:36.43 | c1sco | then this.. |
02:36.43 | c1sco | SIP/100-3805 answered SIP/trxtel.com-3c61f000 |
02:36.43 | c1sco | <PROTECTED> |
02:36.59 | c1sco | what is the native bridge? |
02:37.01 | clyrrad | these all on the same lan? |
02:37.04 | *** join/#asterisk aketchel (n=Eraser@216.189.3.251) |
02:37.04 | c1sco | no |
02:37.17 | enjay- | trxtel.conf is your provider? |
02:37.18 | c1sco | SIP/trxtel.com-3c61f000 is a sip provider |
02:37.20 | enjay- | err.com |
02:37.21 | clyrrad | have you checked the firewalls? |
02:37.41 | c1sco | i am port forwarding 6000-65000 ->> asterisk |
02:37.49 | enjay- | 5060 = SIP |
02:37.50 | c1sco | just to make sure i dont miss any rtp, lol |
02:37.53 | c1sco | and 5060 |
02:37.57 | enjay- | UDP |
02:37.59 | c1sco | yep |
02:38.11 | c1sco | its the response that is not getting there |
02:38.19 | clyrrad | temporarly disable the firewalls on each side - bet its one of them - I had this issue many times where it sits on native bridge - and it was always firewall related |
02:38.20 | c1sco | when i answer the x-lite phone |
02:38.49 | c1sco | clyrrad i only have one firewall, the other side is the provider |
02:39.00 | enjay- | nat involved? |
02:39.06 | c1sco | yes |
02:39.14 | enjay- | nat=yes on the sip device? |
02:39.17 | c1sco | the asterisk box and the x-lite are behind nat |
02:39.24 | enjay- | not that it should matter actually since its local to your server.. |
02:39.31 | c1sco | the asterisk box and the x-lite are on the same lan |
02:39.41 | c1sco | im trying to call from pstn to my xlite |
02:39.51 | clyrrad | then disable it on your side |
02:39.51 | clyrrad | are you behinde a router? |
02:39.51 | clyrrad | with NAT? |
02:39.56 | c1sco | yes look |
02:39.58 | enjay- | SIP trunk have nat=yes? |
02:40.03 | andrejkw | Ok, I have a SIP section called i2telecom.com (don't ask, my provider requires it). And now I am trying to dial through it, but I don't want Asterisk thinkinking that I am trying to dial through the i2telecom.com server. |
02:40.04 | c1sco | sip trunk? |
02:40.12 | andrejkw | How can I come around this? |
02:40.13 | c1sco | i have no sip trunk |
02:40.17 | c1sco | i dont think |
02:40.19 | enjay- | ahh.. |
02:40.25 | enjay- | thought you had a sip trunk to your provider.. |
02:40.38 | clyrrad | so your asterisk is behinde the NAT then |
02:40.40 | c1sco | well, explain how you refer to a trunk in extensions.conf? |
02:40.47 | c1sco | yes, asterisk is behind nat |
02:40.55 | c1sco | and my xlite is behind nat on the same lan |
02:40.57 | enjay- | c1sco from CLI type sip show registry |
02:41.08 | *** join/#asterisk johnnyb (n=jonathan@207.155.33.225) |
02:41.12 | c1sco | enjay- nothing |
02:41.21 | clyrrad | then your xlite is not registered |
02:41.22 | andrejkw | Anyone, please? |
02:41.23 | clyrrad | its not connecting |
02:41.34 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-59-1.cybersurf.com) |
02:41.39 | enjay- | not true.. |
02:41.42 | c1sco | enjay- is is under sip show users |
02:41.52 | c1sco | it is* |
02:42.03 | enjay- | yea thats where xlite would show up.. |
02:42.14 | clyrrad | how bout sip show peers? |
02:42.27 | clyrrad | xlite will show under users instead of peers? |
02:42.29 | c1sco | 100/100 192.168.1.100 D 255.255.255.255 4996 OK (101 ms) |
02:42.29 | c1sco | 2 sip peers [1 online , 1 offline] |
02:43.08 | clyrrad | alright so Asterisk does see the xlite |
02:43.14 | c1sco | i believe so |
02:43.23 | enjay- | I dont wanna lead you down the wrong path cause from my experience you'd have to register with trxtel.com to send/receive "authenticated" calls.. |
02:43.24 | c1sco | yes, i can make outgoing calls from xlite |
02:43.26 | clyrrad | can you dial out using the xlite? |
02:43.48 | clyrrad | pastebin your extensions.conf |
02:43.52 | c1sco | this is what trx telecom says |
02:43.56 | c1sco | You do not need to register with our servers |
02:44.08 | c1sco | clyrrad sorry, how do i pastebin? |
02:44.10 | clyrrad | this is SIP or IAX? |
02:44.11 | c1sco | its a url? |
02:44.13 | c1sco | sip |
02:44.15 | clyrrad | pastebin.com |
02:44.40 | c1sco | clyrrad just warning u my extensions.conf is ghetto, because im just learning |
02:44.53 | c1sco | im trying to bridge the gap from call manager to asterisk |
02:46.21 | c1sco | almost done on pastebin |
02:46.32 | c1sco | its going... |
02:47.12 | enjay- | should give you a url for us.. |
02:47.18 | c1sco | its just hangin |
02:47.27 | c1sco | gotta another way to do this? |
02:47.37 | clyrrad | no |
02:47.41 | c1sco | ok |
02:48.00 | andrejkw | Ok, I have a SIP trunk nam i2telecom.com (don't ask, my provider requires it). And now I am trying to dial through it, but I don't want Asterisk thinkinking that I am trying to dial through the i2telecom.com server. How can I do this? |
02:48.22 | enjay- | eh andrejkw? |
02:48.39 | c1sco | pastebin gave me this error |
02:48.40 | c1sco | Warning: unlink(/home/pastebin/public_html/../cache/recent): No such file or directory in /home/pastebin/lib/pastebin/db.mysql.class.php on line 243 |
02:48.44 | andrejkw | The trunkname is i2telecom.com. For some reason it won't register without it. |
02:48.50 | clyrrad | try pastebin.ca |
02:49.07 | andrejkw | Now I am trying to call through it using Dial, would I do Dial(SIP/0000@i2telecom.com) ? |
02:49.41 | c1sco | http://pastebin.ca/90872 |
02:50.15 | clyrrad | im guessing those first 2 lines are NOT in the file correct? |
02:50.16 | c1sco | i know im doing something horribly wrong... |
02:50.20 | c1sco | correct |
02:50.30 | clyrrad | phew |
02:50.31 | c1sco | well, line 2 is |
02:51.09 | c1sco | im basically focusing on, default1, globals, and pstn-inbound |
02:51.55 | clyrrad | yea your file is a big mess |
02:52.05 | clyrrad | first off your [general] should be at the top of the file |
02:52.44 | c1sco | ok |
02:53.16 | c1sco | the part that im confused about is the USERID part |
02:53.27 | c1sco | and how the ani number is matched |
02:53.38 | c1sco | or if the ani is matche |
02:53.51 | enjay- | did? |
02:54.00 | c1sco | um no |
02:54.12 | Dovid | . |
02:54.12 | c1sco | my number is [globals] |
02:54.12 | c1sco | USERID=9255651913 |
02:54.13 | c1sco | PHONE1=100 |
02:54.13 | c1sco | PHONE1VM=100 |
02:54.13 | c1sco | XLITE=SIP/1000 |
02:54.13 | c1sco | EXTEN=100 |
02:54.15 | c1sco | whops |
02:54.17 | c1sco | sorry |
02:54.20 | Dovid | ~pb |
02:54.21 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
02:54.23 | c1sco | my bumber is 7124325415 |
02:54.34 | c1sco | and my xlite extension is 100 |
02:54.35 | *** join/#asterisk clyrrad1 (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
02:54.36 | Dovid | clsco: please use pastebin |
02:54.42 | Dovid | ~pb |
02:54.43 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
02:54.47 | c1sco | sorry |
02:54.51 | c1sco | accident |
02:55.13 | *** join/#asterisk carrar (i=tim@osburn.com) |
02:55.21 | carrar | anyone using postgres with their asterisk? |
02:55.26 | carrar | I see a -07 in the date |
02:55.29 | carrar | 2006-07-17 13:18:05-07 |
02:55.38 | c1sco | clyrrad any idea? |
02:55.39 | carrar | odd |
02:55.53 | c1sco | carrar its july |
02:55.55 | c1sco | ? |
02:55.57 | carrar | no |
02:56.01 | carrar | after the seconds |
02:56.02 | carrar | see |
02:56.15 | c1sco | milliseconds? |
02:56.23 | carrar | every entry has -07 |
02:56.26 | c1sco | lol |
02:57.12 | carrar | it's not in the master.csv file |
02:57.12 | carrar | just in th db |
02:58.36 | *** join/#asterisk johnnyb (n=jonathan@207.155.33.225) |
02:59.25 | carrar | hahah |
02:59.25 | carrar | oh man |
02:59.36 | carrar | thats funny shit |
02:59.51 | carrar | time zone |
03:00.14 | carrar | I'm just not used to seeing the date like that |
03:00.21 | c1sco | wow |
03:00.23 | c1sco | so im lost |
03:00.33 | carrar | -07 PST8PDT |
03:00.41 | c1sco | my inbound calls are not connecting to my xlite |
03:00.43 | carrar | is what that is |
03:00.55 | carrar | I use xlite |
03:01.00 | c1sco | carrar pst is -8 |
03:01.06 | carrar | not right now |
03:01.09 | carrar | it's -7 |
03:01.11 | c1sco | is it |
03:01.11 | carrar | we change |
03:01.14 | c1sco | ahh |
03:01.29 | c1sco | i use xlite also |
03:01.57 | c1sco | but, when i call from pstn to xlite/asterisk, the pstn side never gets a response back from the asterisk/xlite |
03:02.00 | andrejkw | Ok, I have a SIP trunk name i2telecom.com (don't ask, my provider requires it). And now I am trying to dial through it, but I don't want Asterisk thinkinking that I am trying to dial through the i2telecom.com server. How can I do this? |
03:02.49 | carrar | 1sco |
03:02.52 | carrar | do you Answer it? |
03:02.55 | c1sco | yes |
03:02.56 | c1sco | i do |
03:03.02 | carrar | in the exten |
03:03.04 | c1sco | and the pstn side gives me an error |
03:03.10 | carrar | weird |
03:03.15 | c1sco | cannot connect |
03:03.35 | carrar | none xlite sip work? |
03:03.40 | c1sco | ? |
03:03.47 | c1sco | well, xlite makes outbound calls |
03:03.56 | c1sco | its something wrong i think with my extensions.conf |
03:04.02 | c1sco | http://pastebin.ca/90872 |
03:05.04 | carrar | paste paste |
03:05.04 | carrar | heh |
03:06.15 | *** join/#asterisk clyrrad1 (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
03:06.28 | carrar | which one it taking the call? |
03:06.34 | c1sco | 100 |
03:06.45 | carrar | where is the n,Answer |
03:06.56 | c1sco | ? |
03:07.12 | carrar | try Answer then Dial |
03:07.24 | c1sco | in pstn-inbound? |
03:07.30 | carrar | yeah |
03:07.46 | c1sco | should it be 1,Answer and 2,Dial? |
03:07.56 | carrar | I'm using that |
03:08.00 | carrar | for my xlite |
03:08.05 | andrejkw | Is there any other way how to identify a SIP connection? |
03:08.12 | c1sco | let me try |
03:08.15 | andrejkw | Instead of by the name inside [ ]? |
03:09.25 | c1sco | hey carrar what should USERID be? |
03:09.51 | carrar | whatever number you want to match on |
03:09.58 | c1sco | thats the caller id right? |
03:10.05 | c1sco | it should match all numbers |
03:10.18 | c1sco | inbound caller id numbers right? |
03:10.39 | carrar | use pattern matching |
03:10.53 | carrar | NXXNXXXXXX |
03:10.56 | carrar | NXXXXXX |
03:11.06 | carrar | _NXXXXXX |
03:11.20 | carrar | or . |
03:12.03 | Mr-packet | how can i change what IP address asterisk trys to contact an upstream with.. |
03:12.05 | carrar | andrejkw, why not set a customer caller id name |
03:12.06 | *** part/#asterisk P-NuT (n=P-NuT@CPE-60-227-93-75.nsw.bigpond.net.au) |
03:12.09 | carrar | custome |
03:12.16 | Mr-packet | my asterisk is running on a multi-interface machine |
03:12.20 | andrejkw | would that work? |
03:12.24 | c1sco | Mr-packet externip in sip.conf |
03:12.33 | c1sco | i think |
03:12.44 | Mr-packet | c1sco. thats what i thought |
03:12.57 | Mr-packet | does'nt seem to want to work though |
03:13.09 | c1sco | is it under global? |
03:13.12 | carrar | c1sco, could use the switch option too |
03:13.27 | carrar | if you want a catch all |
03:13.50 | c1sco | hey |
03:14.07 | carrar | I always match the exact number |
03:14.20 | c1sco | when i call from pstn to my xlite, the rtp should go sip proivder --> asterisk box ---> xlite and back right? |
03:14.48 | carrar | pstn->*->xlite? |
03:14.55 | c1sco | no asterisk? |
03:14.59 | andrejkw | carrar: doens't work, I get No route to destination. |
03:15.16 | andrejkw | I need some other way to reffer to the connection in the Dial function. |
03:15.26 | carrar | caller id name |
03:15.30 | *** join/#asterisk topping (n=topping@001-785-676.area1.spcsdns.net) |
03:15.36 | carrar | set it to whatever you want to identify what call it is |
03:15.42 | andrejkw | I can't use the name of it, because it contains a domain, and then it thinks I am trying to call out using that domain. |
03:15.55 | andrejkw | carrar: I just tried, it doesn't work. |
03:16.00 | carrar | it does work |
03:16.26 | c1sco | carrar when i sniff the rtp, i see xlite --> asterisk and thats it |
03:16.31 | carrar | Set(CALLERID(name)=Call From SIP) |
03:16.43 | andrejkw | But you don't understand. |
03:16.48 | andrejkw | That's not what I mean. |
03:16.49 | *** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn) |
03:16.54 | c1sco | this is what i get in console |
03:16.54 | c1sco | -- Executing Dial("SIP/trxtel.com-3c619000", "SIP/100|12|tr") in new stack |
03:16.54 | c1sco | <PROTECTED> |
03:16.54 | c1sco | <PROTECTED> |
03:16.54 | c1sco | <PROTECTED> |
03:16.54 | c1sco | <PROTECTED> |
03:17.06 | carrar | xlite should be talking only to asterisk |
03:17.21 | carrar | is xlite natted? |
03:17.25 | c1sco | do i need stun if im behind nat? |
03:17.32 | carrar | shouldn't |
03:17.33 | andrejkw | I have a SIP provider with [i2something.com] and now I have to reffer to it in Dial(). |
03:17.40 | carrar | as long as your FW is passing it |
03:17.43 | *** part/#asterisk aketchel (n=Eraser@216.189.3.251) |
03:17.52 | andrejkw | When I try to reffer to it, it thinks I actually want to use i2something.com to dial. |
03:17.54 | carrar | sip.conf |
03:17.54 | carrar | nat=1 |
03:17.55 | carrar | ? |
03:18.03 | c1sco | the firewall if forwarding 5060 and 6000-65000 to the asterisk |
03:18.19 | c1sco | sip.conf nat=no |
03:18.30 | carrar | enable nat for your xlite client |
03:18.32 | c1sco | xlite and asterisk are on the same lan |
03:18.34 | carrar | in sip.conf |
03:18.38 | carrar | oh |
03:18.43 | c1sco | both behind nat |
03:19.11 | carrar | can you call your xlite from another xlite? |
03:19.15 | c1sco | yes |
03:19.18 | c1sco | u wanna try |
03:19.23 | carrar | not really |
03:19.24 | c1sco | u can use 101 |
03:19.26 | Mr-packet | i'm slowly going insane |
03:19.32 | c1sco | carrar yes u can |
03:19.41 | c1sco | and i can make outgoing calls |
03:19.48 | c1sco | just this inbound is not working |
03:19.50 | carrar | is your PSTN TDMA? |
03:19.59 | c1sco | TDM? |
03:20.01 | carrar | what is your pstn connection? |
03:20.13 | c1sco | its txr telecom thats all i know |
03:20.21 | carrar | maybe * is trying to bind the two together |
03:20.26 | c1sco | http://www.trxtel.com/ |
03:20.28 | carrar | is it sip? |
03:20.29 | c1sco | i think so |
03:20.36 | c1sco | carrar i think you are right |
03:20.48 | c1sco | its trying to go pstn -> xlite |
03:20.53 | c1sco | and i cant do that |
03:20.58 | *** join/#asterisk pdtmobile (n=ptinsley@c-68-53-40-50.hsd1.tn.comcast.net) |
03:21.00 | c1sco | because my portfowarding is going to asterisk |
03:21.07 | xachen | carrar : trxtel is SIP and IAX |
03:21.15 | xachen | er, c1sco I mean |
03:21.23 | carrar | what is in the console? |
03:21.29 | c1sco | scroll up |
03:21.31 | carrar | does it say it is trying to do that? |
03:21.31 | c1sco | i pasted it |
03:22.07 | carrar | yeah Attempting native bridge |
03:22.12 | c1sco | what is that shit |
03:22.16 | carrar | but does it faile or succed? |
03:22.22 | c1sco | ok |
03:22.24 | c1sco | here is how it goes |
03:22.31 | c1sco | i call from the pstn to my xlite |
03:22.36 | c1sco | xlite rings |
03:22.38 | c1sco | i answer xlite |
03:22.50 | c1sco | the pstn call just keeps ringing |
03:22.59 | c1sco | xlite has dead air |
03:23.15 | c1sco | and pstn call says sorry call cannot be completed |
03:23.19 | carrar | is your pstn g.711? |
03:23.26 | c1sco | not sure |
03:23.55 | carrar | so try putting a xlite client on the internet |
03:23.58 | carrar | see if that works |
03:24.06 | c1sco | ? |
03:24.08 | c1sco | what u mean? |
03:24.12 | carrar | using a internet IP |
03:24.17 | carrar | routable one |
03:24.22 | c1sco | how |
03:24.24 | c1sco | i dont have one |
03:24.42 | c1sco | u think its nat? |
03:24.51 | carrar | could be |
03:24.58 | carrar | could be trying to bridge |
03:24.59 | carrar | OR |
03:25.04 | carrar | codec missmatch |
03:25.19 | carrar | I would think it would say something baout codec though |
03:26.26 | carrar | so enable nat for the hell of it |
03:26.36 | carrar | on sip.conf & xlite |
03:26.37 | c1sco | enable nat? |
03:26.40 | c1sco | heh? |
03:26.48 | c1sco | what do you mean? |
03:26.53 | carrar | nat=1 |
03:26.58 | c1sco | nat=yes? |
03:27.03 | carrar | same thing |
03:27.22 | c1sco | when i do that i think the phone doesnt register |
03:27.32 | carrar | should |
03:27.38 | *** join/#asterisk P-NuT (n=P-Nut@CPE-60-227-93-75.nsw.bigpond.net.au) |
03:27.40 | c1sco | let me try |
03:27.54 | c1sco | it registered |
03:27.59 | c1sco | let me try |
03:28.06 | carrar | I use nat=1 just cause I never know where they are coming from |
03:28.39 | carrar | also what are you allowing in sip.conf for codecs? |
03:28.49 | c1sco | i pickup the xlite and the pstn phone keeps ringing |
03:28.58 | c1sco | carrar all |
03:29.02 | c1sco | i think |
03:29.11 | carrar | try these |
03:29.11 | carrar | disallow=all |
03:29.11 | carrar | allow=ulaw |
03:29.12 | carrar | allow=alaw |
03:29.15 | c1sco | k |
03:29.28 | carrar | althought this sounds like call setup issues |
03:29.47 | c1sco | i agree |
03:29.54 | c1sco | yeah codec did not work |
03:30.02 | c1sco | it really sounds like call setup |
03:30.27 | andrejkw | Is there a way to give a SIP connection 2 identifiers? |
03:30.42 | carrar | c1sco, sets see your sip.conf |
03:30.53 | c1sco | k |
03:31.31 | andrejkw | Come on guys :( |
03:31.34 | c1sco | http://pastebin.ca/90911 |
03:31.35 | andrejkw | There has to be a way. |
03:31.37 | andrejkw | This is crazy. |
03:31.47 | carrar | ok |
03:31.49 | carrar | actaully |
03:31.55 | carrar | back in your extensions.conf |
03:31.59 | c1sco | yeah |
03:32.00 | carrar | where are you calling you xlite phone |
03:32.05 | c1sco | yeah |
03:32.10 | carrar | I see it defined in globals |
03:32.13 | c1sco | umm, pstn-inbound ??? |
03:32.22 | carrar | PHONE1? |
03:32.27 | carrar | == 100 |
03:32.37 | pdtmobile | andrejkw: what do you mean? |
03:32.42 | c1sco | yes |
03:32.51 | c1sco | thats for pstn-inbound |
03:32.55 | c1sco | 2,Dial |
03:32.57 | carrar | replace PHONE1 with c1sco |
03:33.00 | carrar | err |
03:33.06 | carrar | XLITE |
03:33.10 | c1sco | where? |
03:33.13 | c1sco | in globals? |
03:33.17 | c1sco | or pstn-inbound? |
03:33.18 | carrar | line 64 |
03:33.27 | carrar | thats where you want to dial it right? |
03:33.29 | c1sco | what the url? |
03:33.34 | carrar | yeah |
03:33.38 | c1sco | give me the pastebin url |
03:33.40 | carrar | http://pastebin.ca/90872 |
03:33.55 | andrejkw | Well I have a SIP connection and my provider requires that I have "i2telecom.com" as a trunk name (which also ebcomes the identifier). And when I try to dial through it (number@i2telecom.com), Aterisk thinks that I am actually trying to dial through the domain i2telecom.com and not the SIP connection. |
03:34.04 | c1sco | ok PHONE1 should be XLITE? |
03:34.09 | c1sco | line 64? |
03:34.29 | carrar | xlite phone is username 1000 right? |
03:34.33 | c1sco | no |
03:34.37 | carrar | what is it? |
03:35.05 | carrar | You can't dial just 100 |
03:35.07 | c1sco | username im using is extenion 100 |
03:35.09 | Dovid | lol |
03:35.10 | carrar | You can dial SIP/100 |
03:35.15 | c1sco | ok |
03:35.20 | c1sco | so what should i change |
03:35.26 | carrar | change: |
03:35.26 | carrar | # |
03:35.27 | carrar | exten => ${USERID},1,Dial(${PHONE1},30) |
03:35.28 | carrar | with |
03:35.28 | Dovid | clsco: what r u tryin to do ? |
03:35.31 | carrar | # |
03:35.33 | c1sco | in global XLITE=SIP/100 |
03:35.38 | carrar | exten => ${USERID},1,Dial(SIP/100,30) |
03:35.40 | c1sco | Dovid dial inbound from pstn --> xlite |
03:35.44 | carrar | see if that works |
03:35.47 | Dovid | ok |
03:35.50 | c1sco | carrar let me try |
03:36.13 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
03:36.16 | andrejkw | Anyone, please? :'( |
03:36.31 | *** part/#asterisk mog (i=ejabberd@68.62.237.103) |
03:36.33 | carrar | can also change PHONE1=100 to be PHONE1=SIP/100 |
03:37.02 | carrar | also |
03:37.07 | c1sco | also? |
03:37.17 | carrar | I'v never used a variable in the patern maching area |
03:37.28 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
03:37.28 | *** mode/#asterisk [+o mog_home] by ChanServ |
03:37.34 | carrar | exten => 9255651913,1,Dial(SIP/100,30) |
03:37.36 | carrar | use that |
03:37.39 | carrar | just to test |
03:37.40 | c1sco | k |
03:41.10 | c1sco | carrar no luck |
03:41.34 | carrar | what does console say |
03:41.39 | c1sco | same thing |
03:43.36 | carrar | pstn IAX2? |
03:43.57 | c1sco | sip |
03:44.02 | carrar | i don't see it |
03:44.29 | andrejkw | Anyone, please? :'( |
03:44.54 | carrar | You have the contet commented out |
03:45.04 | c1sco | no |
03:45.07 | c1sco | thats another sip account |
03:45.35 | carrar | pstn-inbound is the context in extension.conf right? |
03:45.44 | c1sco | yeah |
03:45.52 | carrar | So where is it in sip.conf? |
03:46.26 | carrar | How is the inbound call getting put into the pstn-inbound context? |
03:46.34 | c1sco | dunno man |
03:46.39 | carrar | heh |
03:46.49 | c1sco | trx telecom sends me the call because i gave them a url |
03:46.56 | c1sco | SIP/100@loke.sciarrilli.com |
03:48.02 | carrar | You need to setup a entry in sip.conf that will take incoming calls from then and put them in the pstn-inbound context |
03:48.09 | carrar | them |
03:48.14 | c1sco | how? |
03:48.19 | carrar | RTFM :) |
03:48.25 | c1sco | which one |
03:48.34 | c1sco | documentation for asterisk doesnt seem the greates |
03:48.39 | carrar | just the sip.conf original file |
03:51.45 | *** join/#asterisk bmg505 (n=leon@c1-114-9.rndf.isadsl.co.za) |
03:51.47 | carrar | I am sure your sip provider can send you a example if they are any good |
03:52.03 | c1sco | what is a good sip provider? |
03:52.06 | c1sco | free one |
03:52.16 | docelmo | good luck |
03:52.34 | docelmo | inexpensive one is plainvoip |
03:52.36 | carrar | free sip is like fre gas |
03:53.12 | *** join/#asterisk ZX81 (n=ZX81@203-173-176-166.bliink.ihug.co.nz) |
03:53.20 | ZX81 | problems with mailing list? |
03:53.30 | ZX81 | Diagnostic-Code: SMTP; 451 4.4.1 reply: read error from lists.digium.com.s8b2.psmtp.com. |
03:53.38 | *** join/#asterisk campbell (n=csteven@ip-202-37-228-10.internet.co.nz) |
03:53.51 | campbell | hi |
03:54.09 | hads|home | hey ZX81 |
03:54.13 | ZX81 | hi |
03:54.13 | ZX81 | :D |
03:54.31 | hads|home | How's tricks? |
03:54.35 | campbell | anyone here running multiple TDM400p cards? if so would you recommend it? |
03:55.10 | hads|home | campbell: Depends on the motherboard. I have one place running two with no trouble. |
03:55.23 | Dovid | campbell: depends on the MB |
03:55.32 | andrejkw | Well I have a SIP connection and my provider requires that I have "i2telecom.com" as a trunk name (which also ebcomes the identifier). And when I try to dial through it (number@i2telecom.com), Asterisk thinks that I am actually trying to dial through the domain i2telecom.com and not the SIP connection. How can I fix this? |
03:55.55 | campbell | yeah that's what i'd read so far, i guess i'll give it a go and see how i get on, cheers |
03:57.06 | ZX81 | hads|home good |
03:57.08 | ZX81 | :D |
03:57.12 | ZX81 | busy but good |
03:57.12 | ZX81 | :D |
03:57.21 | ZX81 | in New Zealand for another two weeks |
03:57.25 | ZX81 | before returning to Italu |
03:57.28 | ZX81 | *italy |
03:57.38 | hads|home | Cool. Having fun over there? |
03:57.45 | ZX81 | yeah, in love!!! |
03:57.46 | ZX81 | :D |
03:57.50 | hads|home | :) |
03:57.54 | ZX81 | campbell: how many is multiple? |
03:59.27 | andrejkw | anyone? please? |
04:00.01 | hads|home | andrejkw: How about Dial(SIP/i2telecom.com/12345) |
04:00.17 | andrejkw | let me try |
04:00.22 | *** join/#asterisk mdiehl (n=mdiehl@c-69-252-219-76.hsd1.nm.comcast.net) |
04:00.40 | mdiehl | Hi all. |
04:00.49 | *** join/#asterisk l0ke (n=loke@c-24-5-215-69.hsd1.ca.comcast.net) |
04:01.02 | l0ke | im back |
04:01.03 | l0ke | its c1sco |
04:01.07 | l0ke | i got dropped |
04:01.10 | l0ke | carrar u here? |
04:01.15 | l0ke | Dovid u here? |
04:01.18 | mdiehl | Anyone gotten Gnomemeeting to work with Asterisk? |
04:01.44 | andrejkw | Nope, didn't work |
04:01.52 | mdiehl | In h.323, that is. |
04:03.48 | *** part/#asterisk ZX81 (n=ZX81@203-173-176-166.bliink.ihug.co.nz) |
04:04.07 | l0ke | i changed password |
04:06.38 | andrejkw | Well I have a SIP connection and my provider requires that I have "i2telecom.com" as a trunk name (which also ebcomes the identifier). And when I try to dial through it (number@i2telecom.com), Asterisk thinks that I am actually trying to dial through the domain i2telecom.com and not the SIP connection. How can I fix this? |
04:08.24 | mdiehl | Has anyone gotten ANY h.323 client to work with Asterisk? |
04:09.39 | hads|home | mdiehl: Doesn't ekiga use SIP these days? |
04:10.21 | mdiehl | Yes, but I'm trying to get some h.323 phones to work and I thought I'd test with an old gnomemeeting client. |
04:10.30 | mdiehl | I'm mostly concerned with getting h.323 working. |
04:11.01 | hads|home | Aha, I know nothing of h323. |
04:13.42 | mdiehl | Bummer. |
04:13.44 | drray | andrewjk , place a [i2telecom.com] section in sip conf? |
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04:57.23 | NoRemorse | hi all |
04:57.33 | NoRemorse | what do I have to do to get * to create auniqueid again please? |
04:57.36 | Dovid | does asterisk from asterisk.org work on FeeBSD ? |
04:59.49 | Qwell | Dovid: yes |
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05:03.04 | NoRemorse | what do I have to do to get * to create auniqueid again please? |
05:03.28 | Dovid | @Qwell for free BSD do i need to do ztdummy ? |
05:03.42 | Qwell | if you want meetme or timing |
05:04.36 | *** part/#asterisk tpak (n=tpak@c-67-190-182-213.hsd1.co.comcast.net) |
05:05.16 | russellb | NoRemorse: it does it automagically |
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05:10.10 | NoRemorse | You have two options in /usr/src/asterisk-addons: |
05:10.10 | NoRemorse | 1. Add a CFLAGS+=-DMYSQL_LOGUNIQUEID to the Makefile. |
05:10.10 | NoRemorse | 2. Add a #define MYSQL_LOGUNIQUEID to the top of the sourcefile. |
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05:17.37 | mdiehl | Anyone gotten Gnomemeeting to work with Asterisk? In h.323, that is. |
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05:52.57 | JT | hmm |
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06:07.11 | FuriousGeorge | i think the switch built into my snom messes with the quality of the call |
06:07.31 | FuriousGeorge | sounds like packet loss |
06:07.38 | FuriousGeorge | smells like it too |
06:11.16 | stoffell_h | FuriousGeorge: are you running so much traffic over the pc ? |
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06:26.57 | docelmo | what does packet loss smell like? Im real curious |
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06:38.35 | negativecreep | hi all |
06:44.27 | docelmo | most are sleeping like where I am going to |
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06:50.23 | diLLec | anyone there ? |
06:51.00 | docelmo | nope.. going to BED |
06:51.33 | diLLec | :) |
06:51.38 | *** join/#asterisk sadiqsb (n=sad@213.132.231.57) |
06:52.17 | sadiqsb | hi can anyone help me setup asterisk for my voip provider |
06:52.25 | sadiqsb | hi can anyone help me setup asterisk for my voip provider |
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06:52.55 | sadiqsb | am using only IPbased SIP trunking |
06:53.15 | sadiqsb | internally i can communicate , i cont call put side |
06:53.16 | negativecreep | sadiqsb: what the problem? |
06:53.32 | sadiqsb | i installed asterisk on fc5 |
06:53.38 | negativecreep | you can't call other sip providers? |
06:53.42 | sadiqsb | internal extention are working fine |
06:53.55 | sadiqsb | i cont call through sip provider |
06:54.07 | sadiqsb | yes |
06:54.14 | sadiqsb | can u help me to set it up |
06:54.18 | negativecreep | which providers? |
06:54.28 | negativecreep | fwd, sipgate? |
06:54.33 | sadiqsb | sip.kingcalls.com |
06:55.18 | negativecreep | have you registered with them? |
06:55.23 | sadiqsb | i thnik some problem with sip.conf /extention.conf |
06:55.24 | sadiqsb | yes |
06:55.29 | negativecreep | as a peer? |
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06:55.37 | sadiqsb | yes |
06:55.39 | negativecreep | they should allow calls from ur sip users. |
06:55.45 | negativecreep | paste ur extensions.conf |
06:55.50 | sadiqsb | k |
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07:00.24 | sadiqsb | hi am back |
07:01.28 | sadiqsb | more extensions.conf |
07:01.28 | sadiqsb | ;KY |
07:01.28 | sadiqsb | ;;;;;;;;;;;;; |
07:01.28 | sadiqsb | [vsky] |
07:01.28 | sadiqsb | exten => 1001,1,Dial(SIP/bbc) |
07:01.29 | sadiqsb | exten => 1002,1,Dial(SIP/test) |
07:01.30 | sadiqsb | exten => 1003,1,Dial(SIP/krish) |
07:01.32 | sadiqsb | exten => 1004,1,Dial(SIP/ravi) |
07:01.34 | sadiqsb | exten => 1005,1,Dial(SIP/sad) |
07:01.36 | sadiqsb | exten => 1006,1,Dial(SIP/zen) |
07:01.38 | sadiqsb | exten => 84.11.110.56,1,Dial(SIP/1001) |
07:01.40 | sadiqsb | ;exten => _0[1-9].,1,Dial(SIP/out/${EXTEN}) |
07:01.42 | sadiqsb | ;exten => _00[1-9].,1,Dial(SIP/out/${EXTEN}) |
07:01.45 | sadiqsb | exten => _9.,1,Dial(SIP/${EXTEN:1}@out,60 |
07:01.46 | sadiqsb | include => default |
07:01.48 | sadiqsb | this is my extentins.conf file |
07:02.01 | my007ms | r_evolution, never POST in the channel |
07:02.15 | my007ms | lol sadiqsb |
07:02.19 | sadiqsb | ok |
07:02.25 | sadiqsb | i have a probel |
07:02.36 | sadiqsb | some one is helping me |
07:02.42 | sadiqsb | i forgort his nick |
07:02.59 | my007ms | u can use http://pastebin.ca/ |
07:03.04 | sadiqsb | i got problem with sip truning with provider |
07:03.23 | hads|home | ~pb |
07:03.30 | jbot | from memory, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
07:03.30 | sadiqsb | can u healp me in this |
07:05.22 | hads|home | sadiqsb: It's rude to PM people. |
07:06.11 | sadiqsb | sorry |
07:06.20 | sadiqsb | can u help please |
07:06.38 | FuriousGeorge | stoffell_h: i know this is a bit late, but its just constant traffic. they use windows remote desktop |
07:10.36 | sadiqsb | <FuriousGeorge> can u help me in sip truning with provider through asterisk |
07:12.53 | sadiqsb | hi can anyone help me Asterisk SIP trunk with VOIP Provider |
07:14.21 | FuriousGeorge | ~docs |
07:14.24 | jbot | docs is probably probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
07:14.31 | hads|home | ~thebook |
07:14.32 | jbot | well, thebook is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
07:15.31 | sadiqsb | i went through docs |
07:15.31 | sadiqsb | internally extention are working |
07:15.46 | sadiqsb | problem for longdistence calls through provider |
07:16.08 | sadiqsb | JBOT---can u help me |
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07:18.04 | sadiqsb | negativecreep --can u help with sip trunk |
07:20.16 | FuriousGeorge | negativecreep is a nirvana song, right |
07:21.10 | hads|home | Yep |
07:21.23 | sadiqsb | FuriousGeorge> he (neg) was helping me he went off |
07:21.37 | sadiqsb | Furi can u help me in sip trunng |
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07:26.44 | negativecreep | back |
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07:31.26 | sadiqsb | <negativecreep> can u hewlp me |
07:31.42 | sadiqsb | in sip truning with provider |
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07:54.01 | CrashHD | can anyone explain why I'm getting 99.975586 during my zttest (this is with a sangoma a104d) and not higher? |
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08:08.11 | CrashHD | *crickets* |
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08:13.51 | Nobbie | heya =) |
08:14.41 | Nobbie | i'm struggling to find documentation on call transferring. currently when users do a blind transfer to an extensions which is busy, the call is dropped. is it possible to make it ring back to the transferee instead of dropping it ? |
08:15.51 | my007ms | Nobbie, make mirco for transferee and add this in it |
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08:18.37 | Nobbie | add which in it ? |
08:18.39 | Nobbie | ;) |
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08:38.35 | Bert- | hello there |
08:39.35 | Bert- | I have a little question : yesterday, all was working fine, and this morning, I'm unable to register to asterisk : 401 unauthorized. I really don't understand why, as I changed nothing :(. Is there a way to find why can't I register to * anymore please ? |
08:42.10 | CrashHD | is 99.975586 acceptable with a sangoma card? |
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08:49.58 | Nobbie | heya =) |
08:50.23 | Nobbie | sorry, got cut off. what was that code i had to add for the macro to not have calls dropped when blindly transferred to a busy extension ? |
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08:55.55 | |oranjia| | hello guys :) |
08:58.04 | Nobbie | hi |
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09:23.26 | smlcoffee | evening |
09:23.34 | Sonderblade | i have a python AGI script which ends with sys.exit(99), but on asterisks console it outputs: AGI Script testagi.py completed, returning 0, strange? |
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09:27.47 | heffernan | hi. |
09:28.04 | heffernan | i want to set CFIM in asterisk with exten => _*21*X.,1,Set(${DB(CFIM/${CALLERIDUM})}=${EXTEN:4}) |
09:28.10 | heffernan | but that does not work. |
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09:28.16 | Strom_C | CFIM? |
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09:28.22 | kindor | sup |
09:28.27 | tparcina | hi channel! |
09:28.35 | kindor | how can i disable the application /system/preferences toolbar? |
09:28.38 | Strom_C | lol internets |
09:28.39 | kindor | or remote it even |
09:28.44 | kindor | oops |
09:28.45 | heffernan | Strom_C: that's not important anyway, what i'm wondering about is how i can set a db value in the dialplan? |
09:28.47 | kindor | sorry |
09:28.52 | heffernan | i can't find anything documented. |
09:28.58 | Strom_C | heffernan: show application set |
09:29.03 | Strom_C | your syntax should be: |
09:29.19 | Strom_C | Set(DB(Family/key)=value) |
09:29.31 | heffernan | okay, no $-s then? |
09:29.43 | |oranjia| | has anyone tried to send faxes over premi cells? |
09:29.49 | tparcina | heffernan: and Set(variabla=DB(family/key)) |
09:29.50 | Strom_C | no, not when you're storing data into a variable or function using Set() |
09:29.57 | Strom_C | tparcina: NO NO NO |
09:30.10 | Strom_C | tparcina: Set(variable=${DB(family/key)}) |
09:30.57 | heffernan | Strom_C: and dbdel? |
09:31.03 | tparcina | strom: ok, i was writing from my head |
09:31.19 | Strom_C | DBDel(family/key) |
09:31.41 | heffernan | Strom_C: okay, so they've kept dbdel. |
09:32.00 | tparcina | Strom: do you know why Set(DB(Family/key)=$VARIABLE) doesn't work? |
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09:32.05 | Strom_C | heffernan: yes, dbdel is still there; dbput and dbget have been deprecated |
09:32.20 | Strom_C | tparcina: because you have to type ${VARIABLE} |
09:32.22 | effectiveape | Howdy. As exepcted still having problems ;) |
09:32.24 | Strom_C | not $VARIABLE |
09:33.43 | tparcina | Strom: I have this - exten => 2,3,Set(DB(forward/${CALLERID(number)})=${FORWARD}) |
09:33.52 | tparcina | and it doesn't work |
09:34.18 | Strom_C | because you're passing the wrong argument to the CALLERID function |
09:34.26 | Strom_C | it's CALLERID(num) |
09:34.35 | effectiveape | bt complaining that the number isn't recognised even though it seems correct |
09:34.35 | Strom_C | show function CALLERID |
09:35.30 | tparcina | o hope that's it. yesterday i have spent two hours trying to figure it out :)) |
09:35.36 | Strom_C | tparcina: try it |
09:36.14 | tparcina | i will as soon as i eat my donuts :)) |
09:37.15 | tparcina | you know every thing has it's priority ;)) |
09:37.42 | Strom_C | so that explains why your emoticons have double chins! |
09:37.54 | Strom_C | :) |
09:38.10 | tparcina | i have exten => 2,2,Eat(cake) and then exten => 2,3, Set(DB... :)) |
09:38.44 | Strom_C | heh |
09:38.47 | Strom_C | mmm, cake |
09:39.20 | *** join/#asterisk Gunnar (n=gunnar@62.97.242.6) |
09:39.34 | tparcina | i don't know why but i like to put double chins :)) it looks bether this way (at lest for me) |
09:39.50 | effectiveape | i was wondering if bt required full msisdn but it still complains :/ |
09:39.52 | *** join/#asterisk Kernel_core (n=I@217.218.80.151) |
09:42.11 | tparcina | effectiveape: bt is british telecom? |
09:42.23 | effectiveape | yeah |
09:42.30 | tparcina | and oyu are using PRI? |
09:42.38 | effectiveape | i'm just getting the 'this number is not recognised' woman |
09:42.44 | effectiveape | bri |
09:43.06 | effectiveape | isdn2e |
09:43.15 | tparcina | well, what do you see on CLI, what number does asterisk dial out? |
09:43.43 | effectiveape | it shows a Dial with the number i've entered (which is my mobile)... 07970xxxxxx |
09:43.59 | negativecreep | if i dial outgoing calls through a provider like fwd, i will be able to meter those calls...or will i lose track of them? |
09:44.11 | Strom_C | negativecreep: what do you mean "meter"? |
09:44.35 | tparcina | <negativecreep: yes, they will be recodred in CDR or database (if oyu use one) |
09:44.36 | negativecreep | Strom_C: calculating their usage..accumulating CDR |
09:45.10 | tparcina | effectiveape: sorry, can't help you out. but maybe you can help me. how can i see who is the owner of one mobile number? |
09:45.35 | negativecreep | What do you recommend as an easy to use solution for prepaid billing. |
09:45.41 | negativecreep | I dont need much fancy stuff though. |
09:45.44 | effectiveape | you mean the provider (ie orange etc...) |
09:46.17 | effectiveape | or the indivdual? |
09:46.24 | tparcina | effectiveape: the individual |
09:46.27 | Strom_C | negativecreep: for postpaid you could just process the CDRs |
09:46.38 | Strom_C | negativecreep: for prepaid, I think there's something called astcc |
09:47.35 | negativecreep | Strom_C: All i need is a solution that can check that if the user connecting has xx number of minutes left to call..if yes, then he can proceed. also, if the number of minutes drop to 0, call should be disconnected. |
09:47.57 | Strom_C | negativecreep: like i said, check out astcc :) |
09:48.07 | effectiveape | Each phone company will have directory enquiriesif you know the provider. |
09:48.34 | tparcina | effectiveape: it starts with +44795....... |
09:48.36 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
09:49.21 | negativecreep | Strom_C: thnx |
09:50.03 | effectiveape | Yeah you can't rely on that since numbers are ported and such. |
09:50.39 | effectiveape | You might try them all - there are only 5 ;) |
09:50.41 | tparcina | effectiveape: what does it mean that number is ported? |
09:50.47 | effectiveape | (or well 6 now) |
09:51.16 | effectiveape | When you are on one contact (eg vodofone) you can move your number to another contract (eg orange) |
09:51.26 | effectiveape | To keep your number |
09:51.37 | tparcina | effectiveape: ok, thank you |
09:51.37 | carl0s- | hmm |
09:51.48 | effectiveape | sorry i couldn't be more help. |
09:51.57 | Strom_C | it's so weird to hear people talk about vodafone again :) |
09:52.02 | effectiveape | heh |
09:52.24 | effectiveape | yep still got the name here ;) |
09:52.25 | Chris-NB | hi |
09:52.25 | carl0s- | I have been getting read to ask questions on least-cost routing UK mobile numbers. Ported numbers are certainly a problem. I wonder if there is a public database of uk mobile numbers and associated GSM operators. |
09:52.26 | tparcina | effectiveape: it realy isn't that inportant anyway. she probably won't call me... :(( |
09:52.39 | Chris-NB | can someone plz look at that: http://pastebin.ca/91121 and can tell me what I'm doing wrong? |
09:52.42 | effectiveape | <PROTECTED> |
09:52.50 | Chris-NB | I'm trieing to connect two asterisk via IAX2 |
09:52.51 | Strom_C | Chris-NB: lemme have a look |
09:53.02 | effectiveape | Can you simulate a dial from the cli? |
09:53.08 | effectiveape | To external |
09:53.22 | Chris-NB | but get err : / |
09:53.23 | Chris-NB | <PROTECTED> |
09:53.23 | Chris-NB | <PROTECTED> |
09:53.42 | Strom_C | because you're using md5 auth on one box and no md5 auth on the second? |
09:53.54 | Strom_C | effectiveape: yes, just type "dial" |
09:54.54 | effectiveape | does that just workj for known extensions? |
09:55.07 | *** join/#asterisk frenzy (n=frenzy@196.46.104.247) |
09:55.15 | *** join/#asterisk appelza (n=pieter@dsl-146-248-226.telkomadsl.co.za) |
09:55.19 | appelza | Hi |
09:55.43 | appelza | What permisions should I change in AMP to allow mp3 file uploads for on hold music? |
09:55.49 | Chris-NB | Strom_C, tried even with md5 auth on both boxes |
09:55.54 | Strom_C | appelza: read the topic and go to #freepbx |
09:56.11 | appelza | ok |
09:56.11 | Chris-NB | when I try to call, i get this IAX debug output: |
09:56.12 | appelza | tnx :p |
09:56.14 | Chris-NB | Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REJECT |
09:56.18 | Strom_C | Chris-NB: and with auth=md5 commented out? |
09:56.23 | Chris-NB | <PROTECTED> |
09:56.23 | Chris-NB | <PROTECTED> |
09:56.28 | Chris-NB | on both boxes |
09:56.32 | Chris-NB | one moment |
09:56.46 | frenzy | hi.. I've got the grandstream bt102. When I hit the hangup for just a tiny second I get a new line while the other goes over to hold... how to do get the line again ? |
09:56.53 | *** join/#asterisk zepmantra (i=what@125.212.110.115) |
09:57.03 | effectiveape | can you show me what i'd type to dial 07970xxxxxx on Zap/g1 ? |
09:57.03 | tparcina | Strom, exten => 2,3,Set(DB(forward/${CALLERID(num)})=${FORWARD}) works, thank you! |
09:57.36 | Chris-NB | same without auth |
09:57.41 | Chris-NB | WARNING[5683]: chan_iax2.c:6986 socket_read: Call rejected by 192.168.68.156: No authority found |
09:57.56 | Chris-NB | but there is an authority! |
09:58.16 | Strom_C | Chris-NB: do an iax2 debug and pastebin the output |
09:58.33 | Strom_C | tparcina: the wonders of reading documentation :) |
09:59.27 | *** join/#asterisk evisu (n=hIRC@bzq-88-153-134-199.red.bezeqint.net) |
09:59.27 | Chris-NB | http://pastebin.ca/91128 |
09:59.47 | Strom_C | oh wait, i'm a moron. try username- instead of user= |
09:59.49 | zepmantra | heelo just wanna ask i have 2fxo, 2fxs , 1 fxs port is not blinking at the back but i have put power on the card for the fxs , on "dmesg" it says "Unable to do INITIAL ProSLIC powerup on module 3 Module 3: FAILED FXS (FCC)" <---is my card destroyed |
09:59.52 | Strom_C | s/-/=/ |
10:00.09 | Chris-NB | Strom_C, on both iax.conf? |
10:00.13 | Strom_C | Chris-NB: yes |
10:00.21 | Chris-NB | just a mom. |
10:00.27 | frenzy | ? |
10:00.28 | *** join/#asterisk darviria (n=dvr@194-105-181-29.ifb.co.uk) |
10:00.29 | Strom_C | zepmantra: how long have you had the card |
10:00.34 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
10:00.35 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
10:00.38 | Strom_C | zepmantra: and it's a tdm400p, right? |
10:00.38 | zepmantra | about 1 week |
10:00.51 | zepmantra | tdm400p clone, openvox a20022p |
10:01.00 | Strom_C | ewwww |
10:01.09 | Strom_C | grind it up and use it as fertilizer |
10:01.14 | Strom_C | then get a real tdm400p |
10:01.16 | Strom_C | :) |
10:01.18 | Chris-NB | Strom_C, same thing! : / |
10:01.32 | Strom_C | Chris-NB: you are doing an iax2 reload after every change, right? |
10:01.38 | zepmantra | Strom_C , my clone card dead? |
10:01.49 | Chris-NB | Strom_C, i dig a reload. do i have to do iax2 reload ? |
10:02.03 | hi365 | Hello! has anyone ever seen this with a sangoma A200? |
10:02.04 | hi365 | ZT_CHANCONFIG failed on channel 3: Invalid argument (22) |
10:02.09 | Strom_C | Chris-NB: iax2 reload will reload just the iax2 configuration |
10:02.26 | Strom_C | hi365: you probably didnt correctly configure zaptel.conf or zapata.conf |
10:02.29 | Chris-NB | Strom_C, and reload the hole thing, so this should do.right ? |
10:02.31 | frenzy | <PROTECTED> |
10:02.39 | Strom_C | frenzy: flash again |
10:02.40 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
10:02.52 | Strom_C | Chris-NB: iax2 reload on both boxes, right? |
10:02.53 | frenzy | huh ? |
10:03.01 | Strom_C | frenzy: a short hangup is called a hookflash |
10:03.08 | Chris-NB | Strom_C, do you know where I can lookup how to configure IAX between two boxes? |
10:03.11 | Strom_C | or just glash for short |
10:03.13 | Strom_C | er, flash |
10:03.13 | Chris-NB | Strom_C, jep, did it on both |
10:03.17 | Strom_C | Chris-NB: hmm |
10:03.25 | frenzy | tried that out |
10:03.27 | Strom_C | do an iax2 debug and pastebin the output |
10:03.31 | frenzy | but I just get another new line |
10:03.40 | Chris-NB | Strom_C, looked at the voip-info and followed that configuration |
10:03.52 | Strom_C | Chris-NB: what version of asterisk are both boxes running? |
10:04.06 | hi365 | Strom_C -> http://pastebin.ca/91132 |
10:04.09 | Zeeek | Chris-NB the iax to iax box thing was discussed in -users - check the archives |
10:04.33 | Chris-NB | Strom_C, iax2 debug lookes like i posted before |
10:04.49 | Chris-NB | Strom_C, Asterisk versions are 1.2.6 and 1.2.7.1 |
10:05.01 | Chris-NB | Zeeek, I'll look at that. thx |
10:05.11 | Strom_C | hi365: are you running asterisk@home? |
10:05.32 | Strom_C | Chris-NB: ok, link me again |
10:05.36 | hi365 | yup. trixbox 1.1.1 |
10:05.46 | Strom_C | hi365: ok, so read the topic of this channel |
10:05.53 | Chris-NB | Strom_C, http://pastebin.ca/91128 |
10:05.56 | hi365 | hu? |
10:06.01 | Strom_C | hi365: you're supposed to go to #freepbx if you're using trixbox |
10:06.19 | hi365 | right, but this is a card issue |
10:06.41 | hi365 | this isnt a matter of configuring extensions or trunks |
10:06.57 | Strom_C | hi365: no, it isnt, but the config is generated by a utility that comes with trixbox |
10:07.15 | Strom_C | so therefore it falls into the trixbox category |
10:07.31 | hi365 | ok. thanks |
10:07.36 | Strom_C | Chris-NB: is that the complete debug output? |
10:08.51 | Chris-NB | Strom_C, thats after the reject: http://pastebin.ca/91138 |
10:09.05 | Chris-NB | Strom_C, but I think it's not from the call |
10:09.44 | Strom_C | Chris-NB: just for kicks, try commenting out everything but type=friend and context=default on both systems |
10:09.48 | Strom_C | Chris-NB: and see if it works |
10:10.19 | Chris-NB | Strom_C, also username and secret ? but I deactevated the guest accound |
10:10.59 | Strom_C | Chris-NB: yes. lacking a username, it will use the entry name |
10:11.21 | Chris-NB | Strom_C, k. I'll try |
10:11.54 | *** join/#asterisk johnnyb (n=jonathan@207.155.33.225) |
10:11.59 | *** join/#asterisk Dibbler (n=Dibbler@zidane.pi-net.net) |
10:12.32 | Chris-NB | Strom_C, same here: http://pastebin.ca/91144 |
10:13.01 | Chris-NB | this is the dial statement: exten => _8XXX,1,Dial(IAX2/tecLas0016/${EXTEN:3},30,r) |
10:13.12 | Strom_C | take off the ,30,r |
10:13.19 | Strom_C | you dont need it in this context |
10:13.53 | E-bola | Hey guys, which ip-adapter would u recomend to for use with asterisk? To connect normal pstn phones to an asterisk sip/iax server |
10:14.08 | pnlarsson | pap2 |
10:14.08 | E-bola | Do you have any prefered models/brands? |
10:14.23 | pnlarsson | linksys/sipura |
10:14.44 | Strom_C | E-bola: digium tdm400p/tdm2400p, digium iaxy, or linksys pap2t are all good products |
10:15.25 | E-bola | what about a spa1001? |
10:15.39 | Strom_C | Chris-NB: E-bola you'll get a lower per-port price with the pap2 |
10:15.41 | Strom_C | er |
10:15.50 | Strom_C | E-bola: you'll get a lower per-port price with the pap2 |
10:16.00 | Chris-NB | Strom_C, same thing : / |
10:16.04 | Strom_C | Chris-NB: im not really sure what the problem is |
10:16.14 | Strom_C | Chris-NB: are these boxes publically accessible from the outside? |
10:16.20 | E-bola | str0m_C: pap2 is quote expensive |
10:16.29 | Strom_C | E-bola: its only $60 last I looked |
10:16.32 | E-bola | costs double the amount of a spa1001, and i only need 1 port |
10:16.43 | E-bola | strom_C: mmm over 100$ here |
10:16.51 | Strom_C | where's "here"? |
10:17.03 | E-bola | denmark |
10:17.07 | Strom_C | ah ok |
10:17.12 | Chris-NB | Strom_C, nop. they are within my lan. connected via the same switch |
10:17.14 | Strom_C | sure, get the spa1001 |
10:17.34 | smlcoffee | we have used the grandstream ht386 for our analog phones and have no problem |
10:17.36 | Strom_C | Chris-NB: is there any way you can forward port 4569 to one of the boxes? i want to see if I can dial it |
10:17.51 | smlcoffee | we cant get the grandstream 2ooo to work |
10:18.09 | Strom_C | grandstream stuff is kind of junky quality |
10:18.29 | E-bola | We gonna try out the linksys spa922 as ip-phone |
10:18.30 | Chris-NB | Strom_C, sry, thats not possible. the firewall infront of me is blocking it |
10:18.36 | E-bola | do anybody havce comments/experiences with those? |
10:18.49 | Strom_C | E-bola: i have an spa942 |
10:18.55 | Strom_C | E-bola: it's not a bad phone |
10:19.16 | Strom_C | E-bola: there are certainly better phones out there, but the spa942 I like for a cheaper phone |
10:19.23 | Strom_C | Chris-NB: you cant open the port? |
10:19.45 | Chris-NB | Strom_C, nop, it's not my firewall : / |
10:19.46 | E-bola | strom_C: i tihnk the 922 is an even me basic/cheap model |
10:19.59 | Chris-NB | Strom_C, here is the iax debug output of the calling box: http://pastebin.ca/91156 |
10:20.06 | Chris-NB | Strom_C, with two different call statements |
10:20.24 | Strom_C | E-bola: same phone sans line keys |
10:20.24 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) |
10:20.37 | E-bola | ok |
10:22.06 | heffernan | hm, if i do: |
10:22.06 | heffernan | exten => s,1,Set(hidden=${DB(HIDENUM/${CALLERIDNUM})}) |
10:22.06 | heffernan | exten => s,n,GotoIf(${hidden}?hidden:nohidden) |
10:22.06 | heffernan | exten => s,n(hidden),SIPAddHeader(Remote-Party-ID: <sip:${CALLERIDNUM}@area7.appsvrslip11.prigw.com>\;party=calling\;privacy=full\;screen=yes) |
10:22.10 | heffernan | exten => s,n(nohidden),SIPAddHeader(Remote-Party-ID: <sip:${CALLERIDNUM}@area7.appsvrslip11.prigw.com>\;party=calling\;privacy=off\;screen=no) |
10:22.13 | heffernan | ups, maybe i should have pastebinned. |
10:22.17 | Strom_C | heffernan: PASTEBIN |
10:22.21 | heffernan | sorry, sorry. |
10:22.25 | Strom_C | grrrrrr |
10:22.32 | Zeeek | fl00d |
10:22.43 | *** part/#asterisk smlcoffee (n=kvirc@adsl-69-106-203-69.dsl.irvnca.pacbell.net) |
10:22.47 | heffernan | anyway, if i do that, it will do both Sipaddheaders. why? |
10:23.02 | heffernan | i only want it to do one of them, depending on the hidden variable. |
10:23.06 | Strom_C | heffernan: because it's falling through the dialplan priorities |
10:23.38 | heffernan | i thought it would only do the one corresponding to the value. |
10:23.42 | E-bola | hanks for the input strom_c |
10:23.48 | Strom_C | heffernan: why not have a label after the second sipaddheader statement that the first one does a goto() to |
10:23.50 | heffernan | (hidden) for hidden=1 |
10:23.52 | Strom_C | heffernan: no |
10:24.03 | heffernan | and (nohidden for hidden="" |
10:24.03 | Strom_C | it goes to that priority and then continues execution |
10:24.20 | heffernan | how can i achieve what i want? |
10:24.27 | Strom_C | i just told you |
10:24.32 | heffernan | ah, there. |
10:24.39 | Strom_C | yay reading |
10:24.41 | heffernan | thanks. |
10:24.57 | heffernan | how do i add a label? |
10:25.13 | Strom_C | the same way you added the other two labels |
10:25.20 | heffernan | oh, okay. |
10:25.21 | heffernan | thanks. |
10:27.38 | heffernan | Strom_C: so like this? http://pastebin.ca/91164 |
10:28.09 | Strom_C | why do you have cont,1? |
10:28.17 | Strom_C | a priority cant have a priority |
10:28.48 | Strom_C | and you shouldnt have two priorities labelled hidden |
10:29.01 | heffernan | just trying to read the docs on voip-info. |
10:29.03 | Zeeek | as long as one is always hidden... |
10:29.07 | Strom_C | exten => s,n(hidden),addheader |
10:29.14 | Strom_C | exten => s,n,goto |
10:29.20 | Strom_C | exten => s,n(nothidden),addheader |
10:29.27 | Strom_C | exten => s,n(cont),whatever |
10:29.44 | Strom_C | otherwise the goto statement that goes to hidden will get confused |
10:29.46 | heffernan | goto(cont) you mean. |
10:29.48 | heffernan | ? |
10:29.55 | Strom_C | heffernan: its pseudocode |
10:30.05 | heffernan | anyway, looks like it worked now. |
10:30.08 | heffernan | thanks a bunch Strom_C. |
10:30.15 | Strom_C | anytime |
10:30.39 | Zeeek | Strom_C now that you solved that, how abput working on world peace? |
10:30.56 | Strom_C | that one is easy |
10:31.05 | Zeeek | kill all humans? |
10:31.09 | Strom_C | kill everyone off |
10:31.11 | Strom_C | yes, exactly |
10:31.21 | Zeeek | yeah, ya see, great minds and all that :) |
10:31.33 | Zeeek | but... |
10:31.58 | Zeeek | what all humans were dead and an asterisk went into a deadlock condition just as several other pbx were calling it? |
10:32.27 | Zeeek | a) if there were no answer() would it ring? |
10:32.40 | Zeeek | b) if there were no 'r', would it answer ? |
10:32.43 | Strom_C | this is like philosophy class gone horribly horribly wrong |
10:33.17 | Zeeek | how many deadlocked sip channels in a martian asterisk install? |
10:33.27 | Zeeek | see it's way too hot to work today |
10:33.49 | Zeeek | or even be serious |
10:33.55 | Chris-NB | Strom_C, I know what the problem is : / |
10:34.04 | Strom_C | Chris-NB: what is it |
10:34.10 | Zeeek | I recommend Crucial for buying memory. They really did a great job |
10:34.18 | *** join/#asterisk johnnyb (n=jonathan@207.155.33.225) |
10:34.18 | Chris-NB | Strom_C, 1. it was a type *grrrrrr 2. is it possible/not possible to #include another file into iax.conf ? |
10:34.44 | Chris-NB | Strom_C, I generate a file out of a database and include it into iax.conf (did it successfully with sip.conf) |
10:34.56 | Strom_C | Chris-NB: I don't understand your first comment |
10:35.06 | Chris-NB | Strom_C, #include "path/to/file.conf" <-- that correct ? |
10:35.24 | Chris-NB | Strom_C, *grrr again a typo!! I misstyped the path to the file |
10:35.43 | Strom_C | Chris-NB: wait wait wait, you're doing this with includes? |
10:35.56 | Chris-NB | Strom_C, jep. thats not good ? |
10:36.13 | Zeeek | it's ok if you reload after changes |
10:36.29 | Strom_C | i don't think you can't do includes with iax.conf |
10:36.30 | Chris-NB | Zeeek, i'm doing that |
10:36.39 | Chris-NB | Strom_C, I'm not sure |
10:36.40 | Zeeek | Strom_C why not? |
10:36.47 | Strom_C | *shrug* just a hunch |
10:37.02 | Chris-NB | Strom_C, It works within sip.conf. just assumed it works in iax. too? |
10:37.04 | Zeeek | easy enough to try Chris-NB |
10:37.15 | Zeeek | let us know, I'd be curious |
10:37.17 | Strom_C | Chris-NB: try this |
10:37.28 | Strom_C | Chris-NB: type iax2 show peers at the CLI |
10:37.36 | Strom_C | do the included sections show up? |
10:37.44 | Chris-NB | jep, nothing shows up |
10:37.53 | Strom_C | well, then that settles it :) |
10:38.11 | Zeeek | I object strenuously |
10:38.12 | Strom_C | no includes! |
10:38.38 | Strom_C | Zeeek: it's half past three, so I'm looking for the simplest solution :) |
10:40.20 | Chris-NB | wait. it works : D |
10:40.32 | Chris-NB | think I've to go back to school and learn to type |
10:40.33 | Chris-NB | *grml |
10:40.39 | Strom_C | what was the typo? |
10:40.42 | Zeeek | <PROTECTED> |
10:40.48 | Zeeek | it reads the file |
10:41.03 | *** join/#asterisk _MDC_ (n=marcus@c-6efde255.06-72-6c6b7013.cust.bredbandsbolaget.se) |
10:41.04 | Strom_C | Chris-NB: what was the typo? |
10:41.28 | Chris-NB | *damned !!! #inclue |
10:41.33 | Chris-NB | I'm going for a break |
10:41.35 | Strom_C | oh |
10:41.44 | Strom_C | yeah, well if you fuck up the include then it won't be included |
10:42.17 | Chris-NB | Strom_C, anyway, thanks for the support! |
10:42.33 | Chris-NB | next time i read the section, and again, and again, then I ask you ; ) |
10:43.13 | Zeeek | it would appear that the chan_iax2 reads the included file but doesn't parse it into the peers |
10:43.31 | Zeeek | I can't imagine why not |
10:44.43 | _MDC_ | Hi all, i'm trying to get asterisk to connect to a h323 gatekeeper, but I can't get it to regsiter my h323 login, is it the alias setting in oh323.conf that sets ther login? |
10:45.11 | Strom_C | h323: the h stands for headache |
10:46.00 | _MDC_ | Strom_C, hehe, but I have no other options unfortunatly |
10:46.22 | Strom_C | _MDC_: I have no experience with h323 |
10:46.56 | Dr-Linux|work | anyone knows any good mp3 musiconhold link? or free prompt messages site? |
10:47.23 | Strom_C | Dr-Linux|work: I'll make recordings for cheap! |
10:48.15 | Dr-Linux|work | Strom_C: we also use our own recordings, even we have an complete IVR team |
10:48.23 | Dr-Linux|work | but i need some personally |
10:49.09 | *** join/#asterisk Goni (n=blah@217.17.247.70) |
10:49.18 | Goni | hello friends :) |
10:49.56 | Goni | I am facing a problem with my ata-188 .. it was authenticating fine earlier but suddenly it not authentication anymore. Giving me SIP/2.0 401 Unauthorized all the times |
10:50.03 | Zeeek | join #myspace |
10:50.11 | Strom_C | hahahahahahahahahah |
10:50.15 | *** join/#asterisk Modcuts (n=bob@lan.proporta.com) |
10:50.22 | Strom_C | Goni: what did you change |
10:50.39 | Goni | I think I just reset the ata nothing else |
10:50.49 | Strom_C | Goni: are the credentials correct in the ata? |
10:50.53 | Goni | 7940 connects fine, but 188 .. it keeps on saying unauthorized |
10:51.05 | Goni | yes, i reset .. re-created accounts many times |
10:52.39 | Goni | any idea? |
10:53.03 | Strom_C | *shrug* |
10:53.13 | *** part/#asterisk tparcina (n=tparcina@lns02-1506.dsl.iskon.hr) |
10:53.34 | Goni | yea, me too .. trying to get it fixed for like 2 days :S |
10:53.46 | Strom_C | try a sip debug |
10:53.54 | Strom_C | see if the device is passing the correct credentials |
10:54.46 | Goni | Jul 18 16:32:54 VERBOSE[2042] logger.c: Transmitting (NAT) to 203.81.198.157:2020: |
10:54.46 | Goni | SIP/2.0 401 Unauthorized |
10:54.46 | Goni | Via: SIP/2.0/UDP 192.168.1.100:2020;received=203.81.198.157 |
10:55.02 | Goni | credentials seems to be ok |
10:55.14 | Goni | From: <sip:8449347@sip.digitallinx.com;user=phone>;tag=2735791333 |
10:55.20 | Goni | To: <sip:8449347@sip.digitallinx.com;user=phone>;tag=as0e31350c |
10:55.24 | Goni | is this ok? |
10:55.29 | Goni | user=phone ? |
10:56.07 | Goni | same acoun its working fine with x-lite |
10:56.45 | Zeeek | Goni is the Grandstream product? |
10:56.58 | Zeeek | s/the/this a/ |
10:59.01 | Goni | no, this is Cisco ata-188 |
10:59.08 | Goni | running trixbox |
10:59.31 | Strom_C | oh good god, not trixbox |
10:59.38 | Goni | Asterisk |
11:01.55 | Bert- | erf |
11:01.58 | Bert- | trixbox :) |
11:02.13 | Strom_C | i mean, seriously, it's right there in the topic |
11:03.02 | mitcheloc | Strom_C, i don't know, the topic isn't exactly very clear, also on some irc clients it's not easy to read.... |
11:03.19 | Goni | just trying to get help ":) |
11:03.42 | Strom_C | mitcheloc: when you join the channel, the topic is printed out in full right there in the client |
11:04.33 | mitcheloc | Strom_C, yes, but it's just not, easy to read you know? i'm thinking big red letters would do the trick ;) |
11:04.44 | mitcheloc | Strom_C, or better yet, some ascii art |
11:05.07 | Strom_C | mitcheloc: no no, we need a sound file that goes "ATTENCION! ATTENCION! 29453 29453 14452 14452..." |
11:05.25 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
11:05.36 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
11:05.50 | Zeeek | oh, there's a topic? |
11:06.06 | Strom_C | yes |
11:06.06 | Zeeek | join #pr0n |
11:06.07 | mitcheloc | Strom_C, you just jinxed the entire channel...thanks |
11:06.18 | Strom_C | ? |
11:06.39 | _MDC_ | When doing Dial(), does asterisk do the call self or does it redirect the client to the new destination? |
11:07.10 | Strom_C | _MDC_: huh? |
11:07.28 | mitcheloc | Strom_C, whoops, i thought you meant "4 8 15 16 23 42" for some reason |
11:07.37 | mitcheloc | oops... now i jinxed it, great |
11:08.24 | _MDC_ | well, I'm trying to get asterisk to act as a proxy between SIP and H323, so when I call SIP/555 asterisk would call OH323/*externalnumber* with the registred gatekeeper |
11:09.32 | Strom_C | _MDC_: that doesnt explain your question though |
11:09.47 | mitcheloc | Strom_C, that's a lost reference for you, in case you don't follow the show |
11:10.02 | Strom_C | what show |
11:10.08 | mitcheloc | Lost |
11:10.18 | Strom_C | huh? |
11:10.19 | mitcheloc | errr -- http://thelostnumbers.blogspot.com/' |
11:11.08 | mitcheloc | supposedly they are jinxed, and bad things happen to the people around the person that knows them |
11:11.17 | frenzy | any php gurus around ? |
11:11.21 | mitcheloc | but to the person that knows them really good things happen...just due to bad circumstances |
11:11.25 | mitcheloc | frenzy, #php? |
11:11.26 | Zeeek | be sure to watch http://GoodNightBurbank.com |
11:11.28 | Strom_C | what the hell are you going on about, mitcheloc |
11:11.53 | mitcheloc | Strom_C, nada, it's late here, i'm ranting, neveeerrrrmind ;) |
11:12.12 | Zeeek | anyone here use VoicePulse Connect? |
11:12.23 | Strom_C | what's this lost thing you're talking about? |
11:12.31 | Zeeek | lost virginity |
11:12.36 | Zeeek | not of this channel |
11:12.54 | Chris-NB | Strom_C, with the right words, the incluDe works fine : D can call a phone on box B from a phone on box A and the other side too : D |
11:13.05 | frenzy | I have a PHP statement... $HD_Form -> FieldEditElement ('name, type, name-one'); but it wont work becuase of the hiphen... what is the correct syntax ? |
11:13.30 | Strom_C | frenzy: I think you are in the completely wrong channel |
11:13.34 | Strom_C | this is #asterisk |
11:13.44 | frenzy | lol.. |
11:13.48 | _MDC_ | Strom_C: hmm, I'm doing the Dial(OH323/number) in asterisk, that call would go via the h323 gatekeeper via asterisk or will the client connect to that gateway directly? |
11:13.57 | frenzy | I'm doing some asterisk intergration.. |
11:14.09 | frenzy | oh well will try #php |
11:14.22 | Strom_C | _MDC_: i don't know, can h323 do reinvites? |
11:14.37 | *** join/#asterisk appelza (n=pieter@dsl-146-246-160.telkomadsl.co.za) |
11:14.47 | appelza | what is the filename of the default on hold music called? |
11:15.11 | Strom_C | appelza: there are three of them |
11:15.23 | Strom_C | appelza: and they live in /var/lib/asterisk/mohmp3 |
11:16.07 | appelza | Ok, and when I add a new one, does it have to be owned by asterisk? |
11:16.15 | _MDC_ | Strom_C; no idea what that is, ok this is what the situation looks like; the h323 gatekeeper have a access to PST and my asterisk will be acting as a SIP server, that will forward some numbers to that gatekeeper, with me? |
11:16.50 | Strom_C | _MDC_: if the asterisk box is converting between different signaling protocols, then it must stay in the media path |
11:17.32 | _MDC_ | Strom_C, the problem is that the client is behind NAT and h323 does do well behind that, so therefore I'm using asterisk on the same net as the gatekeeper and connect to asterisk via SIP |
11:17.49 | _MDC_ | Strom_C, what's a media path? |
11:18.24 | Strom_C | "IP Telephony" |
11:18.35 | _MDC_ | Strom_C, sorry about that silly question... |
11:18.35 | Zeeek | Beginning asterisk docs: |
11:18.35 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
11:18.35 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
11:18.35 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
11:18.35 | Zeeek | http://www.asteriskdocs.org |
11:18.42 | mitcheloc | ~book |
11:18.45 | jbot | it has been said that book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
11:18.45 | Strom_C | ~docs |
11:18.47 | jbot | extra, extra, read all about it, docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
11:18.53 | Zeeek | ~flood |
11:18.54 | jbot | i heard flood is spewing loads of output into a channel; *very* rude in most channels and often grounds for banning. If you want to show a lot of output to someone, ask them to join you in #flood and paste the output there. |
11:18.57 | mitcheloc | heh, you think thats enough for the poor guy? |
11:19.18 | Strom_C | no |
11:19.18 | _MDC_ | yes... |
11:19.29 | MrChimpy | floody jbot :) |
11:19.36 | Strom_C | :) |
11:19.42 | _MDC_ | will come back when i've grow up |
11:20.15 | Strom_C | _MDC_: i was half-kidding |
11:20.24 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
11:21.03 | *** join/#asterisk moodperson (n=moodpers@ss13.lb4.ltk.com.ua) |
11:21.05 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B67BC5.pool.t-online.hu) |
11:21.11 | moodperson | hello people |
11:21.22 | Strom_C | people?! where?! |
11:21.35 | Zeeek | only asterbots |
11:21.46 | moodperson | :) |
11:21.53 | Zeeek | with kind suggestions for light reading |
11:22.18 | Strom_C | hey, ive got a two-inch-thick book on DSL that I consider light reading :) |
11:22.19 | moodperson | any body help about symbian voip clients for smartphone ? |
11:22.48 | _MDC_ | Strom_C, hehe, np |
11:29.48 | CMike | HI all |
11:29.58 | Strom_C | hello |
11:30.10 | moodperson | hi |
11:30.16 | CMike | hiyas |
11:31.11 | Strom_C | hookers! |
11:35.30 | *** join/#asterisk RoyK (n=roy@236.84-48-82.nextgentel.com) |
11:37.33 | CMike | anyone know why I cant set the callerid string ? I set the string, but the outgoing call doesn't show it in the sipheaders. |
11:37.38 | *** join/#asterisk RoyK (n=roy@236.84-48-82.nextgentel.com) |
11:37.42 | CMike | the string looks ok in the cdr though |
11:37.54 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
11:38.11 | *** join/#asterisk effectiveape (n=nick@82.153.22.16) |
11:38.12 | Strom_C | CMike: what version of asterisk |
11:38.17 | Strom_C | CMike: what dialplan string |
11:38.25 | Strom_C | CMike: what SIP provider |
11:38.39 | CMike | zaptel card. |
11:39.03 | CMike | Asterisk 1.2.9.1 |
11:39.05 | effectiveape | any bt isdn users? |
11:39.08 | Zeeek | zaptelcard + sipheaders... |
11:39.17 | Strom_C | CMike: what? |
11:39.25 | Strom_C | CMike: what the hell does zaptel have to do with sip? :) |
11:39.34 | CMike | zaptel = no sip provider :) |
11:39.45 | CMike | maybe I was a bit unclear :) |
11:39.55 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
11:39.58 | Strom_C | CMike: ok, so you're dialing out over a zaptel card? |
11:40.12 | Zeeek | CMike give the whole context |
11:40.23 | CMike | in the sip.conf I use callerid= 850122800 but the sipheaders show the username as callerid |
11:40.53 | Strom_C | put the number in angle brackets |
11:40.55 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B67BC5.pool.t-online.hu) |
11:41.05 | Strom_C | callerid="Name"<3115552368? |
11:41.06 | Strom_C | er |
11:41.08 | Strom_C | callerid="Name"<3115552368> |
11:41.12 | CMike | tried that... |
11:41.32 | Zeeek | CMike post the EXACT line in sip.conf, just that line pls |
11:41.33 | CMike | the cdr shows the right callerid in the src-field |
11:41.33 | Strom_C | what are you dialing, and where is caller ID not showing up as expected? |
11:41.51 | CMike | callerid = "test <850122850>" |
11:41.55 | Strom_C | NO |
11:42.00 | Strom_C | look at my example |
11:42.01 | effectiveape | or does anyone know who might so i can watch out for them |
11:42.02 | Strom_C | callerid="Name"<3115552368> |
11:42.07 | CMike | hold |
11:42.17 | Zeeek | spaces are bad in something=somethingelse |
11:44.11 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
11:44.24 | Zeeek | callerid=Joe Blow <2002> |
11:46.20 | CMike | now the line says: callerid="test"<850122850> |
11:46.30 | Strom_C | good |
11:46.45 | Strom_C | CMike: what are you dialing out over, and where is caller id not showing up as expected? |
11:46.58 | CMike | and the outgoing invite = From: 0850122800 <sip:0850122800@ |
11:47.26 | Strom_C | CMike: please answer my question |
11:47.44 | Strom_C | CMike: what is the final termination point of the call? |
11:47.47 | Zeeek | Strom_C it's often been said that quotes aren't good |
11:47.48 | CMike | I'm dialing out over a ZAP channel (E1) |
11:48.12 | CMike | and I have to put the outgoing callerid in the format 850122850 (without the 0 at the start) |
11:48.45 | CMike | I dont want users to be able to set their own caller id. eg. I have to force the callerid before I dial out on the zapchannel |
11:48.52 | CMike | if you understand how I mean.. |
11:49.18 | Strom_C | so you're forcing callerid based on the station placing the call? |
11:49.23 | CMike | since I have about 50000 DIDs... I have to be able to put the callerid in the database for each sip client. |
11:49.29 | CMike | yep . |
11:49.44 | Zeeek | do it with set |
11:50.01 | CMike | I guess I have to do that.. hm .. |
11:50.29 | Strom_C | CMike: so ok, now that you have the callerid string set correctly in sip.conf, does the callerid show up correctly on the called party's telephone set? |
11:50.48 | CMike | I'm also running a very old version of asterisk, and there I could put the callerid field to any value, and that was the value the was used for outgoing calls (without have to use the set variable) |
11:51.32 | CMike | hold. lemme verify the sipheaders against the old asterisk |
11:51.40 | effectiveape | and (i wonder if anyone will answer ;)) does anyone know why this might happen? http://pastebin.ca/91233 (and should i be worried?) |
11:51.45 | Strom_C | CMike: why are you worrying about sip headers |
11:52.00 | Strom_C | CMike: you're wasting your time with sip headers |
11:52.20 | Strom_C | CMike: if you're setting the callerid on asterisk itself, the sip headers from the phone are obviously not going to reflect that |
11:52.34 | Zeeek | effectiveape does it work? |
11:52.35 | CMike | I need the sipheaders to be correct, since some calls are routed to a sip -gw ... and not through my zaptel |
11:52.55 | CMike | and that sip-gw uses the sipheaders for the callerid |
11:53.11 | Strom_C | CMike: lets debug one thing at a time, shall we? |
11:53.20 | effectiveape | Well other ththe problem i'm here for (which i don't know if it's related or not) seems to. |
11:53.25 | CMike | eg. for an anonymous call i use "Anonymous<850122850>" |
11:53.27 | effectiveape | I can call in ok anyway |
11:53.34 | effectiveape | just the calling out which is the problem |
11:53.44 | Zeeek | effectiveape state your case |
11:53.59 | Zeeek | typing a line every few minutes makes it hard to see what the trouble is |
11:54.19 | effectiveape | Sure... |
11:54.21 | effectiveape | Anyway.... |
11:54.51 | CMike | this is how I need the number to look like: From: "134610120" <sip:134610120@ |
11:55.12 | Strom_C | CMike: do you want me to help you debug the callerid problem, or not? |
11:55.13 | CMike | and in that case the callerid field in the db looks like 134610120 |
11:55.35 | CMike | strom: sure.. maybe I misunderstood you, |
11:55.46 | effectiveape | I'm trying to call out (over isdn2e bri using quadBri) my mobile number (07970xxxxxx) which comes up in the cli correctly. However all i get is the BT automated woman telling me the number isn't recognised |
11:55.53 | Strom_C | let me make sure I understand the route your call is taking |
11:56.13 | Zeeek | effectiveape looking at CLI what does the dial say? |
11:56.19 | Strom_C | SIP phone -> Asterisk -> E1 PRI -> PSTN -> Telephone set |
11:56.44 | CMike | in one case yep .. and also: SIP phone -> Asterisk -> SIP gw -> PSTN |
11:57.01 | CMike | and the latest example need the sipheaders to match the caller id.. |
11:57.03 | Strom_C | CMike: ok, well lets debug the first |
11:57.05 | CMike | I think at least. |
11:57.07 | CMike | ok.. |
11:57.15 | Strom_C | ENOUGH WITH THE DAMNED SIP HEADERS ALREADY |
11:57.20 | Zeeek | heh |
11:57.22 | Strom_C | one thing at a time |
11:57.23 | Strom_C | please |
11:57.29 | CMike | sure sure.. sorry :) |
11:57.50 | effectiveape | Executing Dial("SIP/11-94a5", "Zap/g1/07970xxxxxx") in new stack |
11:58.03 | Strom_C | so now, with the callerid= string correct in sip,conf, does callerid show up correctly on a PSTN phone? |
11:58.18 | effectiveape | <PROTECTED> |
11:58.19 | effectiveape | <PROTECTED> |
11:58.19 | CMike | no it doesn't |
11:58.20 | Zeeek | effectiveape and one assumes the number shown to you is the one you expect to reach? |
11:58.25 | effectiveape | yeah |
11:58.30 | Strom_C | CMike: it's a PRI, correct? |
11:58.33 | effectiveape | muppet ;) |
11:58.38 | effectiveape | call me if you like ;) |
11:58.53 | Zeeek | and g1 works for other calls? |
11:59.13 | effectiveape | that's the only one i've actually tried at the moment. |
11:59.20 | CMike | E1, and and the format of the callerid I send to the PRI must be in the format 850122850 (the username of the sip-client is 0850122850) |
11:59.24 | effectiveape | Since that the only line i have available to call into |
11:59.34 | CMike | so if I understand correctly I must remove the 0 in the callerid |
11:59.41 | Zeeek | call infromation or BT or someone |
11:59.47 | effectiveape | Thing is that it's obviously getting futher than asterisk to bt |
12:00.32 | Zeeek | effectiveape backing up, it's obvious that you're tyring to set a bunck of values the zaptel doesn't like |
12:00.49 | *** join/#asterisk ariel_ (n=Ariel@70-46-87-158.ftl.fdn.com) |
12:01.01 | Strom_C | CMike: you're only giving me little bits of information |
12:01.03 | effectiveape | yeah i've only just seen that though - wasn't complaining before. |
12:01.10 | Strom_C | CMike: I fail to see the total picture here |
12:01.15 | CMike | I thought that useing the callerid string in the sip.conf resulted in the callerid on the ZAPchannel (without setting the callerid before the dial command) |
12:01.16 | *** join/#asterisk esculapio__ (n=ESCulapi@200.88.44.66) |
12:01.19 | effectiveape | those values were setup from jugnhanns setup tools |
12:01.30 | CMike | to shorten it down: |
12:01.30 | Strom_C | CMike: pastebin a sip.conf entry and the zaptel dialout extension |
12:01.41 | Zeeek | effectiveape well zaptel isn't pleased wit them :) |
12:01.52 | CMike | k .. hold |
12:02.36 | *** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar) |
12:03.05 | *** join/#asterisk hwt (n=hwt@curb.thorkildssen.com) |
12:03.26 | effectiveape | yeah i guess but as i say - seems to be working |
12:05.17 | CMike | hm.. what now.. did pastebin just quit working? |
12:06.51 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
12:08.03 | *** join/#asterisk Strom_C (n=strom@gateway.digium.com) |
12:08.08 | Strom_C | bloody hell |
12:08.17 | effectiveape | ah that error was just me |
12:08.28 | CMike | :) |
12:08.30 | effectiveape | insmod'ed 1q instead of 1r from bristuff |
12:12.51 | CMike | does pastebin work for u guys ? |
12:13.03 | Strom_C | pastebin.ca |
12:13.52 | CMike | http://pastebin.ca/91256 |
12:14.22 | CMike | shouldn't the callerid in the sip.conf be the callerid that is used for the peer on the outgoing zap? |
12:14.44 | effectiveape | i might just ring bt |
12:14.59 | CMike | or did I miss something.. ? |
12:15.31 | Strom_C | CMike: perhaps the telco doesnt like the callerid you're sending |
12:16.22 | CMike | something like that.. I think I'm sending 0850122850 |
12:16.33 | CMike | lemme se of pri debug shows what I', sending |
12:17.27 | CMike | hm |
12:17.35 | CMike | wait a minute.. this looks right: Presentation: Presentation permitted, user number not screened (0) '850122850' |
12:18.00 | CMike | BRB.. Have to call the telco guys. |
12:18.04 | CMike | thnx ... |
12:18.34 | Strom_C | see, i told you not to go balls-crazy over the sip headers |
12:18.58 | CMike | :) |
12:19.12 | effectiveape | just got the bri debug output - does anything in here look kooky? http://pastebin.ca/91260 |
12:22.23 | *** join/#asterisk clive- (n=pirch@dsl-145-36-132.telkomadsl.co.za) |
12:22.27 | MrChimpy | balls crazy sounds like some ron jeremy porno |
12:23.08 | |oranjia| | has anyone used asterfax? |
12:23.31 | MrChimpy | presumably, yes. |
12:23.58 | MrChimpy | I clearly sorted that query out for him then. |
12:25.32 | MrChimpy | hmm. my now-fastagi app can start 240 sessions in 0.2s, whereas AGI version fell over in heap at 60 cos it was starting a perl for each line |
12:25.43 | *** join/#asterisk PakiPenguin_ (n=uppal@linuxpakistan/admin/pakipenguin) |
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12:27.43 | *** join/#asterisk ionix (n=ionix@p5025-ipbfp01miyazaki.miyazaki.ocn.ne.jp) |
12:28.46 | clive- | mrchimpy , how different is your fastagi script to your agi script ? |
12:29.51 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
12:29.59 | *** join/#asterisk Vec (n=Vector@dsl-146-119-118.telkomadsl.co.za) |
12:30.31 | clive- | howzit vec |
12:31.00 | MrChimpy | different in that it's a tcpip server |
12:31.11 | MrChimpy | I/O is sockets |
12:31.21 | MrChimpy | arguments are passed differently too |
12:31.34 | MrChimpy | and you obviously don't have to deal with the HUP issue |
12:31.46 | clive- | is this for astcc? |
12:31.54 | MrChimpy | astcc? |
12:32.15 | clive- | calling card..uses HUP |
12:32.50 | MrChimpy | most DeadAGI apps should have a HUP handler if they want to do any cleanup when the caller hangs up mid execution |
12:32.58 | clive- | so fastagi can scale much more |
12:33.24 | MrChimpy | yep, should've used fastagi to start with but didn't know it existed |
12:34.37 | clive- | does fastagi end when the call ends, or does it continue like deadagi ? |
12:37.18 | pdtmobile | clive-: when the call ends the socket disconnects, so you can just (depending on the language) handle that happening and close down that session properly |
12:37.44 | *** join/#asterisk pnlarsson (n=niklas@c83-248-2-120.bredband.comhem.se) |
12:37.50 | pdtmobile | aka, do any clean up your script needs |
12:38.19 | *** join/#asterisk qdk_ (n=qdk@213.150.62.19) |
12:40.17 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
12:41.05 | *** join/#asterisk s0lid (n=s0lid@124.6.176.98) |
12:42.04 | znoG | hah, cool, didn't know you could have the "hotline" functionality in a PAP2 |
12:42.13 | znoG | that means no more dialplan configuration in the PAP2 ! |
12:42.36 | *** join/#asterisk gandhijee (n=gandhije@mail.win-ent.com) |
12:42.46 | *** join/#asterisk andrejkw (n=andrejkw@c-71-57-143-216.hsd1.fl.comcast.net) |
12:42.52 | andrejkw | Hey guys. I have a little problem. My provider requires me to make the trunk name of my SIP connection "i2telecom.com". Unfortunately, this name also becomes the identifier for the connection. Now, when I want to dial through it Asterisk think I am trying to dial through the domain "i2telecom.com" and not the actual connection. Is there anything I can do? |
12:43.41 | *** join/#asterisk daysmen3_ (n=primus@host86-143-4-220.range86-143.btcentralplus.com) |
12:44.44 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
12:46.59 | znoG | andrejkw: the trunk name *has* to be i2telecom.com? have you tried using [i2telecom] ? |
12:47.11 | andrejkw | Yes I did. |
12:47.19 | andrejkw | It won't Register if it's antyhing else. |
12:49.41 | pdtmobile | so you are saying Dial(SIP/i2telecom.com/5551234) |
12:49.50 | andrejkw | Yes. |
12:50.17 | andrejkw | I tried both, Dial(SIP/i2telecom.com/5551234) and Dial(SIP/5551234@i2telecom.com). |
12:52.13 | pdtmobile | actually i'm not awake I meant the second one |
12:52.51 | pdtmobile | what host is your sip connection to? |
12:53.02 | andrejkw | The host is i2telecom.com. |
12:53.15 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
12:53.16 | clive- | what is a "NMI"? |
12:53.28 | PakiPenguin_ | <PROTECTED> |
12:53.45 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
12:53.56 | clive- | paki thanks, I am getting tons of NMI error messages when I try modprobe my wcfxo |
12:53.57 | dpryo | oh oh, dazed and confused. |
12:54.10 | clive- | yes, dazed and confused..:(( |
12:54.25 | clive- | any ideas on how to fix this ? |
12:54.44 | dpryo | It happens when the hardware sends a message that the kernel doesn't understand, or handle. |
12:54.56 | mut | anyone here ever used any hyperlink/karlnet wifi router boards and/or AP PLuses or AP1000's?? |
12:55.12 | clive- | that explains the 1000 times a minute it does this... |
12:55.17 | clive- | any way to fix this ? |
12:56.09 | *** join/#asterisk NoRemorse (n=bah@210-84-10-183.dyn.iinet.net.au) |
12:56.18 | clive- | my alternative is to try the new version of ztdummy |
12:56.29 | NoRemorse | hi all, do I need the zaptel libraries and/or libpri if I dont have any physical E1 card in my box? |
12:56.36 | dpryo | clive-: Probably upgrade the kernel too? |
12:57.12 | clive- | dpyro its the new centos 4.3 ... 2.6.9-34.0.2.ELsmp |
12:57.21 | dpryo | heh |
12:57.39 | dpryo | only 8 kernels old then :) |
12:57.47 | NoRemorse | what is ztdummy and do I need it ? |
12:57.50 | pdtmobile | NoRemorse: you need ztdummy for certain things to work |
12:57.51 | dpryo | 2.6.17.6 is the latest. |
12:57.52 | NoRemorse | cos it is loaded |
12:57.56 | clive- | dpyro..lol... |
12:58.13 | NoRemorse | thanks pdtmobile, but do I need libpri or wont ztdummy compile w/o it? |
12:58.15 | clive- | Noremorse you probably need ztdummy |
12:58.32 | pdtmobile | NoRemorse: you don't need libpri |
12:58.41 | NoRemorse | at some stage in the past I have had a digium E1 card in the system but it is gone now |
12:58.41 | clive- | dpyro, I think I am going to try the new ztdummy,....otherwise I am stuck |
12:58.55 | clive- | the old ztdummy sucked |
12:59.17 | pdtmobile | NoRemorse: it won't hurt to have it but you don't need it if you don't have a E1/T1 card |
12:59.24 | *** join/#asterisk Greek-Boy (n=grb@193.220.93.162) |
12:59.44 | NoRemorse | ok thanks |
12:59.51 | *** join/#asterisk brad_mssw (n=brad@216.155.111.10) |
12:59.58 | NoRemorse | now I gotta work out how to un install the zaptel module |
13:00.34 | pdtmobile | huh? |
13:00.43 | NoRemorse | zaptel module is loaded |
13:00.44 | pdtmobile | ztdummy is part of zaptel |
13:00.57 | NoRemorse | oh but will it compile w/o libpri? |
13:01.07 | pdtmobile | ya |
13:01.10 | moodperson | hm gayus where find "full" prise from www.sipnet.net ? |
13:01.14 | NoRemorse | ok thanks |
13:02.46 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
13:05.50 | *** join/#asterisk tamp4x (n=tampon@64.201.13.51) |
13:05.56 | tamp4x | is it possbile to use the t1 card so that the machine is an IAD? |
13:07.19 | pdtmobile | You want it to be Dulles Airport? |
13:07.36 | CMike | :) |
13:08.34 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
13:09.46 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
13:09.48 | Vorondil | c: |
13:10.07 | Vorondil | bah, wrong window |
13:10.19 | jbroome | blasphemy! |
13:11.19 | CMike | probably an symbolic link for /dev/null |
13:11.23 | CMike | -n |
13:13.44 | Vorondil | hehe |
13:14.01 | Vorondil | CMike: that would make my job a little easier, now that i think about it |
13:14.04 | Vorondil | ;) |
13:14.29 | CMike | heh |
13:17.27 | andrejkw | So, can anyone help please? |
13:19.22 | effectiveape | anyone know what "1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) " |
13:19.25 | effectiveape | means? |
13:20.21 | clive- | ok, new bie question ... will typing "svn update" bring me to the latest version ? |
13:21.45 | effectiveape | it will bring the source to the latest version as long as your not switched to a branch |
13:21.45 | *** join/#asterisk saftsack (n=saftsack@p54A7E68B.dip.t-dialin.net) |
13:22.21 | clive- | effectiveape tahnks, ..how do I know if I am switched to a branch ? |
13:22.36 | effectiveape | If you don't know you're probably not ;) |
13:23.34 | clive- | effectiveape I think I may be because I first downlaoded with : svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2.4 |
13:24.39 | effectiveape | aha ok. It's probably still being updated on the 1.2 branch anyway. (i don't know the * svn layout but i know svn) |
13:24.48 | effectiveape | so doing an update you should be cool |
13:25.20 | clive- | I did and its gives me : |
13:25.21 | clive- | At revision 1242. |
13:25.32 | clive- | is revision 1242 the latest ? |
13:25.38 | effectiveape | with no downloads? |
13:26.06 | effectiveape | must be the latest on the branch you're on |
13:26.35 | clive- | seems like I am stuck in my branch... |
13:26.54 | effectiveape | yeah but that's the current branch anyway so you should be cool |
13:26.58 | *** join/#asterisk auralia (n=ehernand@207.71.51.162) |
13:27.17 | effectiveape | unless you don't want to be on 1.2 |
13:27.19 | andrejkw | Oh come on, I am really desperate. |
13:27.33 | effectiveape | kinding. so am i! ;) |
13:27.34 | andrejkw | I've been sitting at this for 2 days now, and I can't figure it out. |
13:27.34 | clive- | mind you, I just found a way to look at the svn, and that is the latest revision, cool |
13:27.49 | clive- | whats it andre..maybe newbie me can help |
13:28.00 | andrejkw | Hey guys. I have a little problem. My provider requires me to make the trunk name of my SIP connection "i2telecom.com". Unfortunately, this name also becomes the identifier for the connection. Now, when I want to dial through it Asterisk think I am trying to dial through the domain "i2telecom.com" and not the actual connection. Is there anything I can do? |
13:28.22 | andrejkw | I tried both, Dial(SIP/i2telecom.com/5551234) and Dial(SIP/5551234@i2telecom.com). |
13:28.40 | andrejkw | Neither one works. |
13:28.51 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
13:30.06 | andrejkw | Nobody? :'( |
13:31.41 | effectiveape | There has to be some isdn people on here surely.... |
13:31.41 | moodperson | andrejkw: where your put sip resource ? |
13:31.48 | clive- | sorry andre no clue here...I am ure someone on this list can help, send out an email |
13:31.58 | andrejkw | moodperson: what do you mean? |
13:32.13 | moodperson | andrejkw: www.sipnet.net? |
13:32.32 | moodperson | andrejkw: who sip provider ? |
13:32.37 | andrejkw | VoiceStick |
13:32.48 | moodperson | plz home page |
13:33.05 | andrejkw | http://www.voicestick.com |
13:33.18 | moodperson | ok thx |
13:34.09 | andrejkw | Host is i2telecom.com, they require me to name the connection i2telecom.com, so it becomes [i2telecom.com]. And now I have no way to reffer to it without Asterisk thinking that I am trying to use the domain and not the connection. |
13:35.29 | *** join/#asterisk heison (n=heison@ns.somanetworks.com) |
13:36.10 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
13:36.32 | heison | does anyone know of a place where I can get a London number delivered via IAX? |
13:38.36 | E-bola | Do anybody know a free sip client that works with asterisk and can call forward, for windows? |
13:43.07 | jbalcomb | E-bola x-lite by eyebeam perhaps may work for you |
13:43.22 | *** join/#asterisk Dovid (n=dovi5988@pool-71-250-2-14.nwrknj.east.verizon.net) |
13:44.19 | Dovid | can anyone help me with polycom paging |
13:44.20 | Dovid | ? |
13:44.35 | E-bola | jbalcomb: it dont work with call forward unless u pay |
13:44.40 | jbalcomb | Dovid you talking about auto-answer? |
13:45.00 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
13:45.13 | *** join/#asterisk foRza (n=tMs@firewall.hikt.no) |
13:45.14 | Dovid | yes |
13:45.38 | Dovid | jbacomb: yes |
13:45.51 | jbalcomb | Dovid: have you seen the page on the wiki that specifically discusses auto-answer on the polycoms for paging? |
13:45.56 | Dovid | yes |
13:46.05 | Dovid | cant figure it out. its all startight forwars |
13:46.11 | jbalcomb | what seems to be the trouble? |
13:46.14 | Dovid | but still having issues. no clear enough |
13:46.26 | Dovid | i have to edit the image that goes on to the phone correct ? |
13:48.16 | *** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju) |
13:48.28 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
13:48.28 | *** mode/#asterisk [+o anthm] by ChanServ |
13:49.26 | Dovid | ? |
13:50.40 | [TK]D-Fender | Dovid : yes. |
13:51.56 | Dovid | one sec, tryin the wiki again |
13:53.24 | *** join/#asterisk trbldwine (n=trbldwin@adam.ur.northwestern.edu) |
13:53.57 | *** join/#asterisk carl0s- (n=carl0s@compsup.demon.co.uk) |
13:54.48 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
13:54.55 | carl0s- | ugh. how bizzare. I just rang my chick through my Cisco 7960 via Asterisk via the TDM400P, and I could hear my own voice coming back really loud and clear after a slight delay. It could have been a problem at her end, or it might be because I initiated the call with Speakerphone turned on the handset. What do ya think? |
13:56.08 | Strom_C | carl0s-: sounds like a far end echo problem |
13:56.27 | Strom_C | carl0s-: turn your echo canceller on :) |
13:57.05 | [TK]D-Fender | Strom_C : He can't cancel, he signeds a long term contract ;) |
13:57.20 | Strom_C | heh |
13:57.47 | carl0s- | Strom_C: where? |
13:57.57 | Strom_C | carl0s-: in zapata.conf |
14:01.55 | CMike | speaking of Cisco .. is there i working sip firmware for the 7971g ? |
14:02.14 | carl0s- | I just made another call and it was the same. Had to turn the volume right down. Could this be showing because I have had the line unplugged from the card all night? I only just plugged the line back in. Does some kind of line-normalisation happen only when the zap modules are loaded or something? |
14:02.37 | Strom_C | carl0s-: turn. on. echo. cancel. |
14:02.46 | carl0s- | OK OK |
14:02.56 | carl0s- | but it was perfect yesterday. I'll check the conf now :D |
14:03.04 | carl0s- | ;) |
14:03.44 | carl0s- | echocancel=yes |
14:03.44 | carl0s- | echocancelwhenbridged=no |
14:03.44 | carl0s- | echotraining=800 |
14:03.52 | carl0s- | it's already on then! |
14:05.00 | Strom_C | well, try reloading the drivers then if you're convinced there's equalization that goes on |
14:05.19 | carl0s- | I'm not convinced of anything, was just wondered if that's how possibly what might happen. |
14:05.21 | Strom_C | or try calling a different number |
14:06.55 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
14:07.10 | carl0s- | I called two different numbers. I've just restarted asterisk and zaptel, I'll see how it is now. |
14:07.57 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
14:08.10 | heison | anyone know about toll free numbers in London? |
14:09.45 | *** join/#asterisk DrMouse (n=noneof@62.164.176.20) |
14:10.50 | carl0s- | heison: what about them? |
14:12.30 | DrMouse | right, i have a problem. this is with a digium TDM400, I have 2 FXO modules and 2 FXS. ignoring the fxs, dialing out on either FXO i cant get the TX level to be around half. I am in the UK on BT and dont know of a milliwatt test tone number, so i am using the simple method of trying to get the VU to register about 50%. |
14:12.55 | heison | carl0s-: i'm looking for IAX delivered numbers in London... do you have someone to recommend? |
14:13.08 | DrMouse | even at +20 gain the tx level is still too low, and people are complaining that the call is too quiet |
14:13.25 | clive- | does anyone know how to use ztdummy with the new rtc stuff in it ? |
14:13.28 | heison | carl0s-: rephrase, i'm looking for a London number delivered via IAX |
14:13.44 | carl0s- | heison: sorry, I can't help. I thought you were going to ask something like "do they start with 0800?" |
14:14.04 | DrMouse | yet it is fine with a std phone |
14:14.16 | DrMouse | i am using a GXP-2000 |
14:14.23 | carl0s- | heison: if it's freephone, it won't be a London number in particular will it, it'd just be a nationwide 0800 or 0500 number I should think. |
14:14.24 | DrMouse | any ideas? |
14:14.28 | *** join/#asterisk Casterman (n=pierre@kasadsyed.net8.nerim.net) |
14:14.43 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
14:15.08 | clive- | ~seen tilghman |
14:15.19 | jbot | clive-: i haven't seen 'tilghman' |
14:15.19 | jbalcomb | DrMouse |
14:15.24 | jbalcomb | DrMouse: who is complaining the call is too quiet? |
14:16.20 | DrMouse | jbalcomb - the ppl on the other end of the phone. and also on another site with similar setup |
14:16.35 | effectiveape | any isdn junkies yet? |
14:16.52 | *** join/#asterisk shadebob (n=chatzill@ll81-144-114-192-81.ll81.iam.net.ma) |
14:16.54 | jbalcomb | DrMouse any connection with people on the calling end using speaker phone? |
14:17.13 | DrMouse | jbalcomb - nope |
14:17.19 | Strom_C | DrMouse: where are you located |
14:17.26 | Strom_C | DrMouse: and who is your telephone company |
14:17.49 | DrMouse | Strom_C - Leeds, UK. Teleco is BT. |
14:17.52 | carl0s- | Strom_C: echo seems fine now I've reset the 7960 phone. |
14:17.56 | jbalcomb | DrMouse: have you confirmed that raising the txgain has made any difference at all? |
14:18.04 | effectiveape | oooh i'm in leeds too ;) |
14:18.15 | Strom_C | DrMouse: how many thousand feet long is your loop? |
14:18.17 | shadebob | hi, I try to write an AGI. I have to execute an application every 30s on a channel, during a communication. I don't known how I can send EXEC application during the dial... Someone can help me |
14:18.40 | *** join/#asterisk cvv (n=cvv@212.8.35.34) |
14:18.53 | cvv | good evening! |
14:19.07 | Strom_C | evening?! it's 7:20 in the morning! :) |
14:19.48 | cvv | at my clock I have 17:20 ;-) |
14:19.52 | DrMouse | jbalcomb - tx gain makes a difference, but when at an acceptable level to the callee it is 'scratchy' (clipping i think) and DTMF digits dont send properly |
14:20.50 | Strom_C | DrMouse: i think setting your txgain at 20dB is bonkers |
14:21.04 | Strom_C | DrMouse: how many thousand feet long is your loop? |
14:21.18 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
14:21.48 | DrMouse | Strom_C - dunno, but we are about a mile from the exchange. i know +20 is bonkers, but even at that im only getting a peak of about 1/3 on ztmonitor |
14:21.58 | cvv | Can anyone talk me what I need for implamentation conference via VoiP? |
14:22.18 | DrMouse | anyone know if you can put a gain or AGC on the RTP stream? |
14:22.19 | tamp4x | is it possbile to use the t1 card so that the machine is an IAD? |
14:22.25 | tamp4x | is=as |
14:22.59 | Strom_C | DrMouse: what kind of telephone sets? |
14:23.38 | DrMouse | Strom_C - Grandstream GXP-2000's, shit just realised ive got a sipura thing somewhere i can test with, brb... |
14:24.28 | shadebob | Can I execute a Dial and immediately after, without wait for hangup, pass to next action in dialplan? |
14:25.02 | [TK]D-Fender | shadebob : Nothing happens DURING a dial. What you'd need to do is take the channel # before you dial and have a background process poll for it constantly till its gone in order to do something like taht. |
14:26.04 | DrMouse | Strom_C - same with sipura (841 maybe? unsure what model) |
14:26.15 | shadebob | hi D-Fender. A background process can be an AGI? |
14:26.22 | *** join/#asterisk iriga (n=adu@LSt-Amand-152-31-4-219.w82-127.abo.wanadoo.fr) |
14:26.37 | effectiveape | does noone use isdn? |
14:26.39 | *** join/#asterisk _problem_ (n=lokesh_k@estrela.nortenet.pt) |
14:26.51 | MikeJ[Laptop] | yes.. lots of people use isdn |
14:26.53 | shadebob | D-Fender : because asterisk wait the end of AGI to pass to next action... |
14:27.09 | Strom_C | DrMouse: if you plug the line directly into an analog phone and make a call, does the caller think the line is too quiet? |
14:27.15 | effectiveape | surely people must have come up with similar problems |
14:27.20 | shadebob | D-Fender : maybe it exist a tip to avoid the waiting of the AGI end? |
14:27.26 | DrMouse | it seams like asterisk is recieving a very low volume rtp stream from the phones |
14:27.27 | carl0s- | effectiveape: yeh lots of people. You can look at Junghanns BRIstuff if you want to use ISDN2e with a HFC-S cheapo PCI card, else it's ISDN30/PRI stuff. |
14:27.46 | DrMouse | Strom_C - no volume is fine from an analogue phone |
14:27.50 | effectiveape | i'm using a quadBri card at the moment. |
14:28.00 | effectiveape | Just noone seems to be able to help me getting it working ;) |
14:28.21 | Strom_C | DrMouse: odd...id call digium support. they're open now. |
14:28.54 | znoG | andrejkw: you'd think Asterisk would check first if a context by that name exists, else dial via DNS |
14:29.03 | znoG | andrejkw: that is, lookup host by DNS and connect directly |
14:29.26 | DrMouse | Strom_C - any chance you got the number? |
14:29.26 | [TK]D-Fender | shadebob : you need to initiate the process to be run in the background before the dial. |
14:29.29 | Strom_C | andrejkw: oh, please, just call the context something else and use username= |
14:29.41 | Strom_C | DrMouse: IAX2/guest@misery.digium.com/s |
14:29.52 | znoG | andrejkw: still, if you dial using Dial(SIP/i2telecom.com/5551234) it should still use the [i2telecom] info to dial out, IIRC |
14:30.11 | shadebob | D-Fender : create a new thread for exemple? I not a coding guru :s |
14:31.03 | effectiveape | and bt don't seem to like using phones anymore. It's email or nothing for their ISDN support :/ Muppets |
14:31.06 | andrejkw | ? |
14:31.18 | Strom_C | DrMouse: or +1 256 428 6000 |
14:31.19 | andrejkw | Hmm |
14:31.21 | andrejkw | Are you sure? |
14:31.51 | *** join/#asterisk anthonyl (n=anthony@office.midphase.com) |
14:33.02 | andrejkw | That means my problem must be somewhere else |
14:33.21 | carl0s- | effectiveape: well, I can't help as I don't have ISDN. I know there are some smart and helpful people on here but you have to be patient and wait for (a)the right people to show up and (b)them to have some spare energy : |
14:33.23 | carl0s- | ) |
14:33.34 | *** join/#asterisk Hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
14:34.12 | effectiveape | yeah but it's know when to keep asking ;) - I've been on here for 24 hours now. |
14:34.18 | carl0s- | lol |
14:34.34 | [TK]D-Fender | shadebob : Something like that. Call an "Exec" that will daemonize and poll * for the channel. |
14:34.54 | effectiveape | They probably heard i was on here and don't come on ;) |
14:35.26 | shadebob | D-Fender : thanks a lot. How I can daemonize an asterisk application? |
14:35.53 | carl0s- | just sent you a /msg effectiveape |
14:36.01 | [TK]D-Fender | shadebob : its not an * application, its one you'll be making yourself. |
14:36.41 | *** join/#asterisk JohnJacob (n=dhorner@pool-71-127-102-43.aubnin.fios.verizon.net) |
14:37.11 | shadebob | Problem is I want to modify register of the FXS port (wctdm) and I cannot write on /dev/zap/xxx through an external application because asterisk use it :( |
14:37.48 | *** join/#asterisk s0lid (n=s0lid@202.71.179.140) |
14:37.49 | _MDC_ | hmm.. I get seg fault when doing exten => 889,1,Dial(OH323/mc:@10.1.210.24/0013111111), what could I be doing wrong? |
14:39.44 | clyrrad1 | Hi Guys - I am having a really strange problem - all phones can dial each other and leave voice mail - only problem is the MWI is not comming on - howerver if each phone dials into Comedian Mail - it will tell them they have a new message - and idea what could cause this? |
14:39.56 | *** join/#asterisk unixgeek (n=unixgeek@216-220-234-197.exploremaine.com) |
14:42.14 | saftsack | hi |
14:42.23 | saftsack | are there voip hardware telephones which can handle video calls? |
14:43.47 | effectiveape | those '24' motorola's look tasty |
14:44.28 | effectiveape | http://broadband.motorola.com/consumers/products/ojo/ |
14:45.01 | Qwell | saftsack: several |
14:45.12 | saftsack | it looks great ;) |
14:45.22 | saftsack | what are the prices of those telephones? ^^ |
14:45.24 | effectiveape | sip supported although i've never tried it |
14:45.32 | Qwell | saftsack: ~$300 |
14:45.48 | saftsack | if i use h 323 with asterisk it would be possible to do video calling with asterisk, or? |
14:45.50 | clyrrad1 | Qwell - do you have any idea what could cause my WMI problem? Been playing around with it for 2 days now :s |
14:46.22 | carl0s- | ugh. that echo is unbearable. |
14:47.06 | carl0s- | it's not exactly echo though, it's my own voice coming out of the earpeice. |
14:47.13 | jbroome | ugh. that echo is unbearable |
14:47.14 | jbroome | ugh. that echo is unbearable |
14:47.21 | *** join/#asterisk ChrisDe3 (n=Chrisde3@port-87-234-141-161.dynamic.qsc.de) |
14:47.32 | carl0s- | :D |
14:47.41 | jbroome | :> |
14:48.35 | ChrisDe3 | one question: If an absolutetimeout is set to 1 hour... and the call is still up in 1hour and 11 minutes... this is correct behaviour? |
14:49.17 | *** join/#asterisk froguz (n=xxxxx@pc-95-155-104-200.cm.vtr.net) |
14:50.09 | *** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju) |
14:51.25 | *** join/#asterisk NoRemorse (n=bah@210-84-10-183.dyn.iinet.net.au) |
14:51.42 | NoRemorse | hi all, does this mean anything to anyone?! handle_request: Unknown SIP command 'SI16384P/2.0' from '202.161.21.211" |
14:53.13 | DrMouse | excuse the language but #%$*! i switched from using GSM to PCMU and sunddenly, ppl can hear me fine! |
14:53.23 | DrMouse | unless its an intermittant thing |
14:53.34 | *** join/#asterisk johnnyb (n=jonathan@adsl-38-9-196.tulsaconnect.com) |
14:54.29 | DrMouse | thank you all for your help anyway. |
14:55.00 | DrMouse | at least i now have an extension for contacting digium support :) |
14:55.01 | [TK]D-Fender | NoRemorse : garbage packet. |
14:55.36 | froguz | i'm not good at english but i'll try to put it simple: i'm trying to make calls to the PSTN using a Micronet FXO gateway (in proxy mode), wich has the extension 1001 configured in line 1 so, when i dial 1001 it gives me PSTN dialtone, but then i dial the phone number and nothing happens, it continue giving dialtone. is this a DTMF detection problem? |
14:55.45 | DrMouse | after all that, im off for a smoke. once again thanks for your help. |
14:58.15 | andrejkw | Can I have custom caller ID? Like when a known number calls the phone will show the name of the person? |
14:58.25 | andrejkw | And I can set the name in some configuration file? |
14:58.35 | andrejkw | Anything like that possible? |
14:59.51 | *** join/#asterisk websae_ (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
15:01.36 | *** join/#asterisk carl0s- (n=carl0s@compsup.demon.co.uk) |
15:02.24 | *** join/#asterisk DarKnesS_WolF (n=wolf@81.10.111.139) |
15:03.06 | *** join/#asterisk Damin (n=damin@nucleus.nacs.net) |
15:03.11 | *** join/#asterisk mfdutra (n=marlon@200.208.130.16) |
15:03.26 | mfdutra | my asterisk doesn't respect the absolute timeout |
15:03.45 | *** join/#asterisk hohum (n=dcorbe@12.195.58.235) |
15:04.25 | NoRemorse | Garbage packet? it keeps happening, always the same |
15:05.51 | *** join/#asterisk dlynes_ (n=dlynes@S0106001217014b92.vc.shawcable.net) |
15:06.00 | mfdutra | show channel Zap/1-1 |
15:06.01 | mfdutra | Elapsed Time: 17h49m35s |
15:06.01 | mfdutra | TIMEOUT(absolute) = 120 |
15:06.31 | ChrisDe3 | yes I have the same problem |
15:06.38 | *** join/#asterisk burnproof (n=jsharryp@210.213.242.145) |
15:06.45 | ChrisDe3 | http://bugs.digium.com/view.php?id=7546 |
15:07.38 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
15:07.54 | *** join/#asterisk razu (n=razu@87-119-182-133.tll.elisa.ee) |
15:08.03 | *** part/#asterisk websae_ (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
15:08.20 | *** join/#asterisk s0lid (n=s0lid@202.73.164.125) |
15:08.54 | *** join/#asterisk dorphalsig (i=Dorphals@pcsp168-254.supercabletv.net.co) |
15:09.02 | dorphalsig | Hey |
15:10.42 | malverian | Anyone had any issues with 1.2.10? I'd like to upgrade to it from 1.2.9.1 to fix some voicemail bugs and the CLI issues. |
15:11.21 | dorphalsig | I am running 1.2.7 but some zap channels will die ocassionally and wont come up unless I restart asterisk |
15:11.29 | NoRemorse | cant get DISA to work, it says unkown application |
15:11.31 | dorphalsig | when I call it sounds as if the extension was busy |
15:11.49 | dorphalsig | and when I pickup the phone I get nothing |
15:12.36 | dorphalsig | NoRemorse... there is a module you have lo load to get DISA working |
15:12.41 | dorphalsig | check the wiki for its name |
15:12.45 | *** join/#asterisk mog (i=ejabberd@68.62.237.103) |
15:12.45 | *** mode/#asterisk [+o mog] by ChanServ |
15:12.47 | NoRemorse | app_disa.so? |
15:12.55 | NoRemorse | i'll put it in modules.conf ty |
15:13.06 | dorphalsig | np |
15:13.17 | burnproof | NoRemorse: check if you have actual app_disa.so on /usr/lib/asterisk/modules |
15:13.33 | dorphalsig | Good Point |
15:13.59 | NoRemorse | yeah its there |
15:14.08 | NoRemorse | and so was noload= |
15:14.10 | NoRemorse | lol |
15:14.12 | burnproof | NoRemorse: show applications like disa ? |
15:14.15 | burnproof | what's the output |
15:14.38 | burnproof | on your CLI show applications like disa ? what's d output |
15:15.37 | NoRemorse | nah think I have to restart, reload doesnt fix it |
15:15.45 | NoRemorse | I'll wait till calls drop |
15:16.31 | jbalcomb | Anyone know fairly well how long your data center will stay up on battery backup? |
15:16.55 | *** join/#asterisk websae (n=websae@adsl-64-149-206-121.dsl.milwwi.sbcglobal.net) |
15:16.55 | NoRemorse | 30 mins to 2 hours |
15:17.00 | jbalcomb | I'm trying to decide how long ours should stay up. I'm think 15 minute minimum pushing for 30 |
15:17.14 | jbalcomb | NoRemorse: roughly how many servers do you have? |
15:17.24 | NoRemorse | long enough to ensure a generator startup :) |
15:17.58 | NoRemorse | what ups u got? |
15:20.19 | andrejkw | Is there anyw ay to have a custom caller ID name for every person that calls? |
15:20.30 | dorphalsig | I am running 1.2.7 but some zap channels will die ocassionally and wont come up unless I restart asterisk. when I call it sounds as if the extension was busy and when I pickup the phone I get nothing |
15:20.55 | [TK]D-Fender | jbalcomb : I am in the planning phases for that here. My take is 1 hour tops on the std servers (file, mail, etc), and 2 hours on phone (switching, etc). |
15:21.43 | *** join/#asterisk DasTech (n=DasTech@c-67-176-28-65.hsd1.co.comcast.net) |
15:21.54 | jbalcomb | NoRemorse: we have 3 triplite 2000s, 2 apc 2200s, and 3 apc 1400s. |
15:21.58 | clyrrad1 | Hi Guys - I am having a really strange problem - all phones can dial each other and leave voice mail - only problem is the MWI is not comming on - howerver if each phone dials into Comedian Mail - it will tell them they have a new message - Asterisk also sends an email to the person saying someone has left a message - yet no MWI is lit up on the phones - I have tried the phones on another server and the MWI does wor |
15:22.02 | DasTech | Gouten Morgen |
15:22.09 | jbalcomb | [TK]D-Fender : have you come up with anything yet? |
15:22.12 | DasTech | need some input for dial logic |
15:22.18 | [TK]D-Fender | clyrrad1 : pastebin your setup and describe your hardware |
15:22.33 | [TK]D-Fender | jbalcomb : not model #'s offhand, but it was a BIG rackmount APC solution. |
15:22.51 | [TK]D-Fender | jbalcomb : 10$K or so |
15:22.53 | jbalcomb | [TK]D-Fender: 1 UPS for each rack or just 1 UPS? |
15:23.18 | jbalcomb | [TK]D-Fender: that's nearly the price of a generator |
15:23.25 | [TK]D-Fender | jbalcomb : Simpler intelligent UPS (networked), with massive battery. |
15:23.33 | DasTech | I am working on extension with web interface I want to put a dial button and have it dial me and then play back PLease hold while I connect your call then dial the other person |
15:23.38 | clyrrad1 | The hardware is are GNET VP104s phones that have been tested on other servers - and have used alot of them - the server is remote / on a different network then the phones - they connect remotely - they are not having firewall issues as I have completely disabled the firewall to trouble shoot that |
15:23.40 | [TK]D-Fender | its the remote start/;stop that also adds to the cost. |
15:24.03 | DasTech | has anyone done this |
15:24.08 | *** join/#asterisk salviadud (n=ralfalfa@dsl-201-128-132-150.prod-infinitum.com.mx) |
15:24.12 | clyrrad1 | TKD-Fender can I PM you any config you want to seee? |
15:24.17 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
15:24.19 | salviadud | pastebin |
15:24.19 | [TK]D-Fender | clyrrad1 : pastebin yoru voicemail / SIP setup |
15:24.53 | clyrrad1 | Voicemail is being left using a Macro and the system is recording and storing the messages becase they can be checked by dialing into comedian mail |
15:25.01 | clyrrad1 | all voice mail stuff is working perfectly just no MWI |
15:25.19 | Dovid | TK: i edited the polycom directory file but for some reason the phone wotn get ti |
15:25.22 | Dovid | or so it seems |
15:25.26 | [TK]D-Fender | clyrrad1 : PASTEBIN. I stop telling me everything is alright when its NOT WORKING. |
15:25.43 | NoRemorse | when I goto DISA it cerates a second CDR with the same uniqueid any way to kill the initial one? |
15:25.45 | clyrrad1 | what would youl ike to see? 1 context and the Macro? |
15:25.51 | dorphalsig | I am running 1.2.7 but some zap channels will die ocassionally and wont come up unless I restart asterisk. when I call it sounds as if the extension was busy and when I pickup the phone I get nothing |
15:25.53 | [TK]D-Fender | Dovid : that was remarkably unclear.... |
15:26.06 | [TK]D-Fender | clyrrad1 : sip.conf and voicemail.conf |
15:26.14 | Dovid | one sec |
15:27.13 | mfdutra | Set(TIMEOUT(absolute)=timeout) cannot have spaces around '=' |
15:27.16 | mfdutra | shit bug |
15:27.32 | mfdutra | no warning, no error |
15:28.07 | NoRemorse | cant.. stay.. awake.... must.. sleep.... |
15:28.50 | *** join/#asterisk kindor (n=roy@office.open-ict.nl) |
15:29.19 | clyrrad1 | Here is the pastebin http://pastebin.ca/91354 |
15:29.57 | Dovid | TK: i edited the MacID-directory.xml and when the phone boots i tell it to provision but whent he phone is on i dont see the numbers in the phone |
15:31.03 | GerbilWrk | Anyone know the keycombination to set the IP of a Linksys SPA1001? |
15:32.17 | *** join/#asterisk enjay- (n=enjay@71.216.165.97) |
15:32.21 | razu | GerbilWrk : pap2 combination should work there ... |
15:32.28 | enjay- | morning |
15:32.36 | clyrrad1 | [TK]D-Fender - did you get the pastebin? |
15:32.37 | froguz | i'm trying to make calls to the PSTN using a Micronet FXO gateway (in proxy mode), wich has the extension 1001 configured in line 1 so, when i dial 1001 it gives me PSTN dialtone, but then i dial the phone number and nothing happens, it continue giving dialtone. is this a DTMF detection problem? |
15:32.53 | GerbilWrk | razu, pap2? |
15:33.38 | *** part/#asterisk mfdutra (n=marlon@200.208.130.16) |
15:35.11 | razu | GerbilWrk : does ****111# work ? |
15:35.18 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
15:36.24 | GerbilWrk | possibly |
15:36.28 | GerbilWrk | it said to enter value |
15:36.57 | razu | GerbilWrk : then check out this manual ... there are some codes you need to know : http://www.freshtel.net/support/hardware/Linksys%20PAP2.pdf |
15:37.36 | razu | GerbilWrk : pap2, spa2100 and spa1001 ivr should use the same codes |
15:37.37 | froguz | GerbilWrk, ****111# your*desired*ip*number# |
15:38.06 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
15:38.11 | froguz | GerbilWrk do you read spanish? |
15:38.32 | salviadud | i read spanish |
15:39.10 | froguz | salviadud, me too. i'm asking to GerbilWrk. |
15:39.22 | clyrrad1 | [TK]D-Fender you still here? |
15:39.29 | salviadud | yeah, pinche cabron, lee español GerbilWrk |
15:39.38 | froguz | hahahaha |
15:39.59 | salviadud | well, would someone be interested in listening to a prank call i made recently? |
15:40.07 | salviadud | i called a rehab clinic in florida |
15:40.19 | salviadud | pretending i was high on something |
15:40.28 | clyrrad1 | lol |
15:41.22 | Qwell | link? :P |
15:41.28 | salviadud | could someone give me a name for this "upload shit" sites? |
15:41.41 | Qwell | google video? |
15:41.42 | salviadud | i just need to upload it somewhere |
15:41.49 | Qwell | oh, audio |
15:41.50 | salviadud | does google video take mp3? |
15:41.57 | froguz | GerbilWrk: 100# say dhcp, 101# set dhcp, 110# say ip, 111# set ip, 120# say mask, 121# set mask, 130# say gateway, 131# set GW, 73738# reset |
15:41.57 | Qwell | doubt it |
15:42.55 | clyrrad1 | I am having a really strange problem - all phones can dial each other and leave voice mail - only problem is the MWI is not comming on - howerver if each phone dials into Comedian Mail - it will tell them they have a new message - Asterisk also sends an email to the person saying someone has left a message - yet no MWI is lit up on the phones - I have tried the phones on another server and the MWI does work - any id |
15:43.21 | *** join/#asterisk boch (n=root@201.216.241.97) |
15:43.33 | *** join/#asterisk dimitrich (n=dimitri@lns-bzn-49f-62-147-167-75.adsl.proxad.net) |
15:45.51 | saftsack | will the future bring mass video telephony? |
15:46.05 | anthonyl | sexxy ness |
15:46.31 | salviadud | http://d.turboupload.com/d/790589/donitadunes.mp3.html |
15:46.59 | dorphalsig | I am running 1.2.7 but some zap channels will die ocassionally and wont come up unless I restart asterisk. when I call it sounds as if the extension was busy and when I pickup the phone I get nothing |
15:47.13 | eKo1 | saftsack: it will bring mass video telephony porno actually |
15:47.30 | salviadud | yeah, that damn porno |
15:47.40 | saftsack | thats right too ;) but do you think normal people will use video telephony? |
15:47.47 | salviadud | i just don't get it... where do they get their money from? |
15:47.56 | salviadud | i'm not buying pr0n... |
15:48.03 | effectiveape | i can't see people using it |
15:48.16 | salviadud | i'm boycotting pr0n, and it doesn't seem to work |
15:48.26 | GerbilWrk | razu, that did it, thanks |
15:48.40 | saftsack | i cant see it too but maybe the reason is why nobody uses it |
15:48.43 | eKo1 | I said for porno. The purpose of VoIP is to make another medium to deliver porno. |
15:48.48 | saftsack | a circle ;) |
15:49.04 | salviadud | video telephony will be availabe when the hardware gets cheap |
15:49.14 | saftsack | yes think so too |
15:49.37 | saftsack | but video telephony can just be there if everybody has a broadband internet connection |
15:49.47 | saftsack | and this means a big upload too |
15:51.28 | eKo1 | salviadud: I think webcams are cheap enough as it is. |
15:51.54 | salviadud | well, you wouldn't expect some dude to be at his computer while using the phone all the time |
15:52.18 | salviadud | you gotta integrate |
15:52.20 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
15:52.28 | *** join/#asterisk s0lid (n=s0lid@210.213.199.63) |
15:52.30 | salviadud | webcam + iaxy |
15:52.35 | salviadud | iaxy-cam phone |
15:52.40 | salviadud | something like that |
15:52.44 | _MDC_ | I keep on getting the following error message when doing a H323 call; H.323 call 'ip$localhost/29214-0c9250b7' cleared, reason 12 (Not enough bandwidth), could it be that asterisk announces the wrong ip? |
15:54.20 | salviadud | hey, i need some feedback with this prank call i made |
15:54.32 | salviadud | mostly, because i recorded it in wav format |
15:54.47 | salviadud | then i used sox to compress it into mp3 |
15:54.56 | *** join/#asterisk DarKnesS_WolF (n=wolf@81.10.111.139) |
15:55.05 | salviadud | i don't have a soundcard available, so i don't know if it sounds ok |
15:55.26 | *** join/#asterisk awe6 (n=lba@user-12lml5g.cable.mindspring.com) |
15:55.36 | salviadud | somegeek, with that said, would anyone like to hear it? |
15:55.46 | salviadud | i meant, so |
15:56.57 | eKo1 | _MDC_: the message clearly states: Not enough bandwidth |
15:57.23 | _MDC_ | eKo1, but asterisk and the gk is on the same lan, 100Mbit |
15:57.44 | *** join/#asterisk Gunnar (n=gunnar@62.97.242.6) |
15:58.11 | *** join/#asterisk bkervaski (n=bkervask@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
15:58.13 | _MDC_ | eko1, using ekiga/ohphone on the same machine agains the gk is working just fine |
15:58.49 | bkervaski | Hi all. Is there a way to get * to display a specific string of text when a call queue is ringing it's extensions? i.e., poke a caller id string in there so the people in the queue know what queue is ringing there phone? |
15:59.12 | eKo1 | then I guess this is another bug in the h.323 asterisk channel driver |
16:00.16 | _MDC_ | eKo1, i'm using the oh driver |
16:01.31 | eKo1 | Good luck then. If you do find a bug, let the developers know. I'm no H.323 expert so I can't help any further. |
16:01.48 | *** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju) |
16:01.55 | clyrrad1 | Can anyone help me out with the MWI problem? |
16:02.02 | [TK]D-Fender | clyrrad1 : You config looks fine |
16:02.26 | clyrrad1 | ok.... thats what I was thinking - so why the MWI problem? Its very strange |
16:02.33 | _MDC_ | eKo1, ok, thanks anyway |
16:02.34 | [TK]D-Fender | clyrrad1 : I'm betting there's something with the phone if it working fine aside from that and you are able to enter you box only to see new messages upon entry |
16:02.42 | clyrrad1 | is there any known bugs with it? |
16:02.46 | dorphalsig | I am running 1.2.7 but some zap channels will die ocassionally and wont come up unless I restart asterisk and when I pickup the phone I get nothing, when I call it sounds as if the extension was busy, and when I look at the cli I see the extension is onhook but has an owner. is there anyway to fix this? |
16:03.10 | clyrrad1 | TKD - the phones work on the other Asterisk server |
16:03.15 | eKo1 | dorphalsig: Post it as a bug. |
16:03.42 | clyrrad1 | I am using Asterisk 1.2.9.1 |
16:04.09 | bkervaski | Hi all. Is there a way to get * to display a specific string of text when a call queue is ringing it's extensions? i.e., poke a caller id string in there so the people in the queue know what queue is ringing there phone? |
16:04.48 | droops | hey bkervaski, where are you trying to display it, in the cli? |
16:05.21 | bkervaski | No, on the SIP phone's caller id display.. right now it shows the callerid of the person calling, I was hoping to have it display the name of the queue... |
16:05.27 | eKo1 | Has anybody tried the chan_ss7 driver? |
16:06.27 | dorphalsig | I am running 1.2.7 but some zap channels will die ocassionally and wont come up unless I restart asterisk and when I pickup the phone I get nothing, when I call it sounds as if the extension was busy, and when I look at the cli I see the extension is onhook but has an owner. is there anyway to fix this? Or at least to restart the channel without having to restart asterisk? |
16:07.32 | droops | bkervaski, have you played with Set(CALLERID) |
16:07.37 | *** join/#asterisk saftsack (n=saftsack@p54A7E68B.dip.t-dialin.net) |
16:07.58 | bkervaski | Nope. So you're suggesting handle it in extensions.conf? That's a good idea. I'll give it a whack. |
16:08.11 | droops | thats the first thing i would try |
16:08.40 | droops | no idea if that will do what you need |
16:08.44 | droops | =o) |
16:09.33 | bkervaski | What's the format? set(callerid|mystring) ?? |
16:10.04 | [TK]D-Fender | clyrrad1 : Wish I could give you some more insight..... |
16:10.07 | clyrrad1 | [TK]D-Fender - any ohter idea what can be the problem? |
16:10.12 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
16:10.13 | *** join/#asterisk tdonahue (n=tdonahue@207.138.151.58) |
16:10.17 | clyrrad1 | hrm - :( This is so frustrating |
16:11.05 | droops | Set(CALLERID(NUM)=123456767) |
16:11.14 | [TK]D-Fender | clyrrad1 : If calls flow, then MWI indication should as well. |
16:11.19 | droops | or |
16:11.24 | droops | Set(CALLERID(number)=123456767) |
16:12.02 | droops | thats how i notify a office cell phone, that the call is comming from the office, and isnt a personal call |
16:12.56 | [TK]D-Fender | droops : I have a magic 867-5309 dial option for comic releif ;) |
16:13.17 | clyrrad1 | TKD - yea thats why I am so confused by this - do you know of any bugs with MWI on this version? |
16:13.28 | droops | that will be implemented today |
16:13.34 | droops | awesome idea |
16:13.49 | tzafrir_laptop | clyrrad1, #asterisk-dev is not #asterisk 2-tier support |
16:14.27 | file | unless we're in a nice mood |
16:14.33 | file | or waiting for a build to finish |
16:15.40 | clyrrad1 | sorry - just at a loss on this one i been messin with it for 2 days now |
16:16.42 | clyrrad1 | file with the sip debug show "MWI"? |
16:16.54 | clyrrad1 | will* |
16:17.16 | file | it should show chan_sip sending a packet to your phone saying, "this is the current state of voicemail... this is how many new, and how many old" |
16:17.34 | clyrrad1 | yea i dont see any of that |
16:17.48 | *** join/#asterisk bjohnson_ (n=bjohnson@jecinc.tor.istop.com) |
16:17.58 | file | when leaving a voicemail? |
16:18.25 | clyrrad1 | have not tried while leaving voice mail - is that info not sent all the time? |
16:18.33 | clyrrad1 | I will try now while leaving voicemail |
16:18.49 | eKo1 | ~centosbug |
16:18.50 | jbot | hmm... centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
16:20.38 | clyrrad1 | file - there is no such message being sent while leaving a message too.... |
16:21.16 | stoffell_h | clyrrad1: and what does "show voicemail users" show in the cli? (to be sure the message is left correctly) |
16:22.23 | clyrrad1 | it shows that that there are messsages I can pastebin if you like its 5 lines |
16:22.47 | stoffell_h | we believe you :) |
16:23.27 | clyrrad1 | http://pastebin.ca/91416 |
16:23.36 | *** join/#asterisk RoyK (n=roy@193.75.62.110) |
16:23.37 | clyrrad1 | haha - Ok |
16:24.27 | *** join/#asterisk CoffeeIV (i=rgr@cpe-70-112-100-20.austin.res.rr.com) |
16:24.35 | stoffell_h | clyrrad1: what type of phones are they? |
16:25.18 | stoffell_h | RoyK: on the ferry? ;) |
16:25.52 | RoyK | stoffell_h: i'm off the ferry.... |
16:25.57 | clyrrad1 | They are GNET VP104S phones |
16:26.55 | stoffell_h | clyrrad1: no experience on those, but as file says, you should see MWI messages with tcpdump or sip debug... (if your phone subscribes to MWI that is) |
16:27.29 | clyrrad1 | yes the phones do work with MWI as I had said before they work on another asterisk server |
16:28.00 | stoffell_h | clyrrad1: ok, then try with tcpdump or sip debug to see what message you get "on the other" server, and see if you get these on "this" server |
16:28.10 | stoffell_h | there 'has' to be a difference in config.. |
16:28.38 | clyrrad1 | Well is the MWI not set in sip.conf with mailbox=? |
16:28.46 | wunderkin | um i assume that someone has checked voicemail.conf and sip.conf? |
16:29.23 | clyrrad1 | Yes many times here is the config http://pastebin.ca/91354 |
16:29.26 | wunderkin | clyrrad1, are you specifying the context |
16:29.53 | wunderkin | clyrrad1, invalid id |
16:30.35 | wunderkin | must have just expired |
16:30.44 | *** join/#asterisk lunk (n=lunk@66.152.8.184) |
16:31.07 | lunk | is there a way to set the Ttr dial flags when using .call files? |
16:31.15 | clyrrad1 | I will paste it again |
16:31.51 | clyrrad1 | here it is http://pastebin.ca/91426 |
16:32.03 | *** join/#asterisk nortex (n=breeves@snapper.titanspecialties.com) |
16:32.51 | *** join/#asterisk McLazarus (n=mcallist@72.78.49.117) |
16:32.55 | stoffell_h | clyrrad1: your "other" server, same version as your current? |
16:33.14 | clyrrad1 | no different versions |
16:33.31 | McLazarus | Hi, I am sure I am missing something obvious in the docs but is there a way to take the output of a "pri debug span n" and just have it written to a file instead of the console? |
16:33.41 | McLazarus | or at least have it do both? |
16:34.35 | clyrrad1 | the working server is using HEAD and the non working one is using 1.2.9.1 |
16:35.12 | RoyK | clyrrad1: you mean trunk? |
16:35.28 | stoffell_h | i'd say: compare sip debug.. to be sure they both sent the same info.. |
16:35.36 | clyrrad1 | I got it from CVS head awhile back |
16:35.52 | clyrrad1 | stofeell_h - I see no MWI info being sent |
16:36.16 | stoffell_h | clyrrad1: and do you see it being sent on the working server? |
16:37.06 | clyrrad1 | I have not checked - but it seems pretty clear it is being sent on that server since the MWI works on that server |
16:37.38 | stoffell_h | clyrrad1: try sip debug on current server and restart a phone, the phone should at least "try" to subscribe to MWI... |
16:37.42 | froguz | somebody has worked with micronet fxo gateways?? |
16:37.47 | stoffell_h | (sip debug or tcpdump ..) |
16:39.41 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
16:40.16 | RoyK | stoffell_h: sip debug is usually good for a start |
16:40.54 | clyrrad1 | stofell_h yes i have tried to restart the phones and watch the sip debug i dont see any MWI stuff |
16:41.21 | stoffell_h | clyrrad1: ok, are you in the possibility to try exactly the same on the other server? |
16:41.37 | clyrrad1 | mean setup wise? |
16:41.54 | stoffell_h | no, just booting the phone and making sure you see anything on MWI passing by.. |
16:42.13 | clyrrad1 | yes - I have done that - on the other server the MWI works |
16:42.18 | *** part/#asterisk iriga (n=adu@LSt-Amand-152-31-4-219.w82-127.abo.wanadoo.fr) |
16:42.27 | McLazarus | ah, stupid. I see I just have to do "pri set debug file" |
16:42.28 | stoffell_h | clyrrad1: so you see mwi messages when using sip debug? |
16:42.39 | clyrrad1 | I see the MWI light turn on the phone |
16:42.48 | stoffell_h | McLazarus: thanks for sharing, i didn't knew that either! |
16:43.02 | stoffell_h | clyrrad1: ok, but you didn't check the sip debug then.. |
16:43.13 | lunk | does anyone know how to set trunk options from inside a .call file? |
16:43.22 | clyrrad1 | nope - becase it was working on the other server there are so many phones it will fly by - i wouldnt see it anyway |
16:43.54 | stoffell_h | hm, that's why I prefer tcpdump :) you can select info from specific ip's |
16:44.32 | *** join/#asterisk paryl (n=chatzill@209.236.78.59) |
16:44.34 | Dr-Linux|work | hi guys |
16:44.35 | clyrrad1 | Yea - this is so strange - the phones work in every other regard - just not the MWI |
16:44.39 | Dr-Linux|work | stoffell_h: hey |
16:44.39 | clyrrad1 | it makes no sense |
16:44.52 | clyrrad1 | Hi Dr-Linux |
16:45.01 | stoffell_h | hey Dr-Linux|work |
16:45.10 | clyrrad1 | Everyone who has looked at my pastebin configs has found no issue with them.... |
16:45.21 | paryl | clyrrad1: what model of phone? |
16:45.25 | stoffell_h | clyrrad1: it doesn't make sens now, but if you can, try to compare, make some tcpdumps |
16:45.27 | Dr-Linux|work | well, i just saw someone is using those entries for a user in sip.conf: |
16:45.28 | Dr-Linux|work | callgroup=1 |
16:45.29 | Dr-Linux|work | pickupgroup=1 |
16:45.29 | Dr-Linux|work | call-limit=1 |
16:45.34 | stoffell_h | there 'has' to be a difference in info being exchanged.. |
16:45.42 | Dr-Linux|work | anybody can guide me, what's those for? |
16:45.56 | clyrrad1 | they only thing I can think is that they are different versions of Asterisk |
16:46.14 | clyrrad1 | I dont know if there are bugs with my current version - but it was stable so I doubt it is a bug |
16:46.40 | stoffell_h | clyrrad1: yes, and then it would be great to know if the info being exchanged is the same or not. it could still be a bug, or something else that's being overlooked |
16:46.53 | stoffell_h | Dr-Linux|work: http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups |
16:47.11 | paryl | i'm having issues with IAX calls going dead, sometimes in one direction, sometimes in both, for about 10 seconds. the calls resume like normal after that. it's totally random, and there are no errors in the logs... i can't figure out what could be causing it? |
16:47.23 | Dr-Linux|work | stoffell_h: ok, and any clue about "call-limit=1" ? |
16:47.47 | stoffell_h | Dr-Linux|work: check it on voip-info, i blieve it's something with max. calls being sent to a device |
16:47.53 | clyrrad1 | stofell_h - yea not sure how else to troubleshoot this - and if i do a show voicemail users it shows that there are new messages waiting |
16:48.21 | Dr-Linux|work | stoffell_h: i google it, but coudn't find, so came here |
16:49.24 | stoffell_h | clyrrad1: i vote for `tcpdump host <ip>` ... :) |
16:49.48 | clyrrad1 | what do you want me to look for? |
16:50.08 | clyrrad1 | I can only tcp dump on the server running asterisk though |
16:50.10 | stoffell_h | clyrrad1: and use ethereal to cmopare the 2 streams.. (ideal to see if MWI info is exchanged while booting the phone) |
16:50.11 | *** join/#asterisk visba (n=dca[lapt@sta-208-139-193-162.rockynet.com) |
16:50.17 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
16:50.37 | stoffell_h | clyrrad1: that's good, you see all info going from/to the asterisk server and your phone |
16:51.05 | RoyK | clyrrad1: that's ok. tcpdump -s0 -w somefile udp and host x.x.x.x |
16:51.06 | RoyK | for instance |
16:51.14 | *** join/#asterisk Tako-san (n=Tako-san@24.108.162.254) |
16:52.27 | clyrrad1 | ok doing that what should i grep the file for? |
16:53.00 | stoffell_h | clyrrad1: use ethereal on your workstation to open the capture file.. it's easier |
16:53.08 | stoffell_h | or should I say wireshark... :) |
16:53.24 | RoyK | #include <pcap.h> |
16:53.25 | RoyK | := |
16:53.27 | RoyK | :P |
16:53.36 | stoffell_h | lol |
16:53.44 | clyrrad1 | well not too famaliar with etherreal - but i do have the dump in the file and I can see it - just need to know what to look for |
16:54.30 | stoffell_h | clyrrad1: the mwi info... (assuming you did this on the 'working' server) |
16:55.00 | clyrrad1 | do you know if the dump will actually say 'mwi' or something else? |
16:55.28 | stoffell_h | that I don't know for sure, with ethereal you might be able to "follow the stream" |
16:55.54 | stoffell_h | clyrrad1: can u put the dump file online? |
16:56.06 | ChrisDe3 | to use the function "call-limit" ... I will have to add an entry in sip.conf (respectively a column in the database), right? and set call-limit=1 e.g.? |
16:56.48 | *** join/#asterisk think_ (i=think@noguff.net) |
16:56.58 | ChrisDe3 | because... it doesn't work for me |
16:57.00 | paryl | what can i do if an FXO card doesn't seem to be getting hangups very quickly? |
16:57.13 | clyrrad1 | im checkign the dump file now |
16:58.56 | rob0 | paryl: wouldn't happen to be an X10[01]P, would it? |
16:59.07 | *** join/#asterisk Delta239 (n=adfadsf@200.124.18.171) |
16:59.17 | paryl | rob0, it's a sangoma a200 |
16:59.17 | Delta239 | hey does the sendmail service affects asterisk? |
16:59.30 | stoffell_h | Delta239: n.o.t. |
16:59.45 | rob0 | Ah, in that case I have no idea. |
16:59.52 | [TK]D-Fender | paryl : Need to try and get your telco to enable disconnect supervision |
16:59.59 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
17:00.04 | clyrrad1 | Ok checked the dump file on the working server and see nothing about MWI.... |
17:00.05 | [TK]D-Fender | paryl : Either polarity reveral or cut. |
17:00.08 | think_ | anyone know how to "busy out" a pots line through asterisk? |
17:00.43 | rob0 | Delta239: if voice mail is to be emailed, you need a sendmail binary. Doesn't need to be a full MTA, could be something like nullmailer. |
17:00.46 | paryl | [TK]D-Fender: k, i'll take a look at that, thanks man |
17:01.12 | burnproof | paryl: if either of the two won't work on your end you could fiddle on busydetect parameter on zapata.conf |
17:01.21 | Delta239 | the main problem is that i can't get asterisk to run |
17:01.24 | ChrisDe3 | anyone familar to call-limits? |
17:01.41 | Delta239 | while is booting up it says nAsterisk service started but then when i type in asterisk -r |
17:01.42 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
17:01.55 | Dovid | TK: the file format changed for the pollycoms, since the page on the wiki for auto answer was created |
17:01.57 | Delta239 | it comes with the error message saying that asteriskctl exist? |
17:02.02 | Dovid | any ideas ? |
17:03.33 | *** join/#asterisk Bullseye_Network (n=info@216.143.192.69) |
17:03.59 | *** join/#asterisk johnnyb (n=jonathan@adsl-38-9-196.tulsaconnect.com) |
17:06.27 | clyrrad1 | stofell_h on the working server I just found this... |
17:06.28 | clyrrad1 | Messages-Waiting: no |
17:06.28 | clyrrad1 | Message-Account: sip:asterisk@ |
17:06.28 | clyrrad1 | Voice-Message: 0/5 (0/0) |
17:06.35 | clyrrad1 | is that what you were looking for? |
17:07.11 | clyrrad1 | becase if so - that information is not being sent on the NON working server |
17:08.23 | stoffell_h | clyrrad1: hm, okay |
17:08.57 | clyrrad1 | that give any clues? |
17:08.57 | stoffell_h | clyrrad1: on cli; do "show modules", is "app_hasnewvoicemail.so" and "app_voicemail.so" loaded? |
17:09.46 | clyrrad1 | on the working server yes |
17:09.49 | clyrrad1 | checking the non working one |
17:10.23 | clyrrad1 | yes they both seem to be there on both servers |
17:10.31 | clyrrad1 | the all have a Use Count of 0 |
17:11.00 | clyrrad1 | they* |
17:11.05 | stoffell_h | hm, okay, then i'mout of thoughts.. |
17:11.22 | clyrrad1 | haha - yea thats what i keep runnign into - its so frustrating |
17:11.22 | stoffell_h | unless you can try up- and/or downgrading.. |
17:11.40 | clyrrad1 | I still wonder if its a bug |
17:11.42 | stoffell_h | to see if it's version-related |
17:12.08 | clyrrad1 | guess I will have to try to upgrade :s |
17:12.56 | Dovid | can anyone help me with polycom paging ? |
17:12.59 | stoffell_h | yes, see if it stays.. (still I wonder if the phone subscribes correctly to the MWI) |
17:14.14 | clyrrad1 | it must since they work on the other server |
17:14.31 | Cresl1n | maybe the MWI tube is full |
17:14.33 | stoffell_h | to be sure, tcpdump could tell you .. |
17:14.51 | clyrrad1 | Cresl1n - how could i determine that? |
17:15.00 | *** join/#asterisk lung (n=lung@24-148-96-186.ip.mhcable.com) |
17:15.11 | Cresl1n | just look |
17:15.18 | Cresl1n | make sure there aren't any inflatable dolls in it |
17:15.18 | clyrrad1 | look where? |
17:15.27 | clyrrad1 | very funny |
17:15.28 | stoffell_h | omg... |
17:15.36 | stoffell_h | ;) |
17:15.39 | eKo1 | I just configured my te410p and zttool is telling me they're all OK. shouldn't they be all on red though? |
17:15.41 | Cresl1n | they seems to clog up the tubes a lot ;-) |
17:16.25 | clyrrad1 | looks like gonna have to try the upgrade route |
17:17.02 | clyrrad1 | no one can find config probs in this channel or in the dev chanel so must be version related.... |
17:17.43 | stoffell_h | clyrrad1: indeed, tcpdump should show if the MWI of the phone subscribes though...:) |
17:17.54 | clyrrad1 | I have tried it and found nothing |
17:18.09 | clyrrad1 | both on the working server and the non working one |
17:18.17 | clyrrad1 | I have only found it with sip debug on the working server |
17:18.26 | clyrrad1 | the non working server is not sending any such messages |
17:18.47 | stoffell_h | hm, oke.. i thought the phone had to "subscribe" to * to see the messages |
17:20.08 | lung | does anyone know if this is a valid invite: "INVITE sip:1##########;npdi=yes@x.x.x.x SIP/2.0" |
17:20.18 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
17:20.41 | *** join/#asterisk RoyK (n=roy@193.75.62.110) |
17:22.51 | lung | it works in pre-1.2.7, but not after due to bug 6409 |
17:27.43 | *** join/#asterisk bitboy (n=amit@adsl-065-012-197-229.sip.bct.bellsouth.net) |
17:28.56 | bitboy | hello. |
17:30.31 | bitboy | Anyone know if following is possible: dial an extension---this dials a number, once that call is over, another number is autodialed. So I dont want the channel closed after first call |
17:33.25 | bitboy | anyone? |
17:42.37 | andrejkw | I am having hard time getting FWD to work. |
17:42.43 | andrejkw | Outgoing calls just don't work. |
17:42.56 | andrejkw | I get that it's circuit-busy all the time |
17:43.33 | *** join/#asterisk rene- (n=rene-@dsl-200-67-175-250.prod-empresarial.com.mx) |
17:44.13 | rene- | can PauseQueueMember be used from the CLI or the Manager interfase? because i can see it in show applications but i cannot execute it from the cli |
17:44.26 | rene- | or is it something i can accomplish via the asterisk db? |
17:45.35 | rob0 | andrejkw: I can only share my own recent experience with FWD. SIP does not work at all, which might be because the FWD clock was about 6 minutes off the last time I called 613. IAX2 works fine. Also ... |
17:46.15 | rene- | the command is QueuePause |
17:46.31 | rob0 | ... I tried to sign up at their forums, but the autoresponder was broken, so I could not get the confirmation. "Bad address syntax," is what my Postfix said. |
17:46.33 | andrejkw | I can't get IAX2 to rgister. |
17:46.45 | andrejkw | It just says Rejected. |
17:46.45 | *** join/#asterisk auralia (n=ehernand@207.71.51.162) |
17:46.50 | rob0 | did you just now sign up? |
17:46.59 | andrejkw | Yes |
17:47.05 | andrejkw | well 3 - 4 hours ago |
17:47.13 | auralia | should there be an alarm in zttool if there is no phone line connected to the tdm and the PSTN? |
17:47.21 | rob0 | I think it took about that long for it to register for me. |
17:47.30 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
17:47.38 | andrejkw | So shoild I try again and see? |
17:47.43 | andrejkw | *should |
17:48.00 | MrChimpy | .- |
17:48.08 | rob0 | Follow the instructions for FWD IAX2 at the wiki, and yes, try again. |
17:48.44 | rob0 | fwd-out/783889 192.246.69.186 (S) 255.255.255.255 4569 OK (40 ms) |
17:49.43 | *** join/#asterisk auralia_aNew (n=none@207.71.51.162) |
17:50.26 | auralia_aNew | sorry i got kicked off, i just asked if zttool should have any alarms if there is no line connecting my tdm2400 and the PSTN |
17:51.04 | *** join/#asterisk oceanlan|dustin (n=info@cpe-24-210-253-66.woh.res.rr.com) |
17:52.08 | andrejkw | Do I absolutely need 2 entires (one for incoming and one for outgoing)? |
17:52.10 | *** join/#asterisk anonymouz666 (n=anonymou@20151155235.user.veloxzone.com.br) |
17:52.39 | *** join/#asterisk Iam8up|lpy (n=iam8up@cpe-24-210-253-66.woh.res.rr.com) |
17:53.00 | anonymouz666 | anyone use Dell PowerEdge 850 in here? |
17:53.06 | rene- | how do i send the parameters for Asterisk Manager's QueuePause |
17:53.08 | rene- | ? |
17:53.21 | rene- | dial plan application works like (queuename|interfase) |
17:53.34 | rene- | but i dont know what the right syntax would be for AMI |
17:53.55 | *** join/#asterisk dwmw2_gone (n=dwmw2@baythorne.infradead.org) |
17:54.23 | *** join/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca) |
17:54.45 | oceanlan|dustin | Are T1 "smartjack" interfaces symmetric? Can I replace a telco smartjack with Asterisk and a quad-T1 card? |
17:55.20 | andrejkw | I still get Rejected :( |
17:57.44 | *** join/#asterisk bitboy (n=amit@adsl-065-012-197-229.sip.bct.bellsouth.net) |
17:58.59 | *** join/#asterisk klictel (n=klictel@207.107.208.137) |
18:03.39 | *** join/#asterisk ayamkeren (n=makoata@bb219-74-196-86.singnet.com.sg) |
18:05.19 | *** join/#asterisk saftsack (n=saftsack@p54A7E68B.dip.t-dialin.net) |
18:08.03 | my007ms | hello any one use E1 there |
18:08.45 | my007ms | after i start asterisk with E1 card from digium i get in asterisk CLI |
18:08.54 | my007ms | this error |
18:09.14 | my007ms | chan_zap.c:6337 handle_init_event: Detected alarm on channel 14: No Alarm |
18:09.14 | my007ms | Jul 18 14:04:35 WARNING[11738]: chan_zap.c:1435 zt_disable_ec: Unable to disable echo cancellation on channel 14 |
18:09.19 | rob0 | andrejkw: the advantage of 2 entries vs. a single "friend" entry is that you can use different contexts for incoming and outgoing calls. Seems important to me, but YMMV. |
18:09.21 | my007ms | for all chanels |
18:10.17 | andrejkw | ook |
18:10.40 | ayamkeren | i just finished installing fedora core 5, but when i try to compile zaptel, it's failed, any idea how to do it correctly ? |
18:12.00 | jbroome | you have the build tools installed? |
18:13.13 | Dovid | my polycom all of a sudden has been taking for ever to boot after provisioning and it wont contact the FTP server anymore. anyonehave an idea ? |
18:14.06 | Dr-Linux|work | question, is there any command on CLI> could tell me that how many users are registered with my asterisk? |
18:14.23 | *** join/#asterisk unit (n=doom@Toronto-HSE-ppp3780161.sympatico.ca) |
18:14.25 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
18:14.26 | *** part/#asterisk ChrisDe3 (n=Chrisde3@port-87-234-141-161.dynamic.qsc.de) |
18:15.36 | clive- | did Nufone go bankrupt? |
18:15.36 | *** join/#asterisk Coeus (n=Coeus@ip24-255-125-43.dc.dc.cox.net) |
18:15.39 | my007ms | Dr-Linux|work, yes there is |
18:15.56 | Dr-Linux|work | my007ms: what's that? |
18:16.03 | my007ms | Dr-Linux|work, if u speek sip |
18:16.06 | rob0 | Dr-Linux|work: perhaps "sip show peers" and "iax2 show peers", whatever other protocols you might be using. |
18:16.42 | TrixVox | clive-: Yes |
18:17.08 | clive- | Trixvox wow, ...what a pity |
18:17.36 | gandhijee | anyone here use a TDM400 in a RISC machine? |
18:17.40 | Dr-Linux|work | rob0: that doesn't show as i want |
18:17.42 | TrixVox | not really |
18:18.01 | clive- | Trixvox, I wonder what my balance was...:( |
18:18.44 | andrejkw | Anyone here using FWD with IAX. I can't get mine to register, it's been 4 hours since I enabled IAX support on the account and I am still getting "Rejected". Is there something else I need to do? |
18:19.28 | Damin | TrixVox: They are up and running.. w/ new partners.. NuFone isn't dead.. |
18:19.43 | Damin | TrixVox: My 800 number stil works, and my other one is being ported... |
18:20.40 | clive- | one thing I can say about Nufone, is that he was always honest in the dealings I had with him, I do hope that they are up and running still |
18:21.20 | clive- | damin can one still get in touch with them ? |
18:21.35 | Damin | clive-: Yep.. support@nufone.net. (or is it .com?) |
18:21.50 | clive- | damin thanks |
18:21.54 | Damin | clive-: Your balance is probably still there.. go login and check.. |
18:22.23 | clive- | my balance was always screwed up...:) |
18:23.54 | *** join/#asterisk apocn (n=apo@225stb46.codetel.net.do) |
18:24.50 | apocn | Can I originate a call from one extension to another from the CLI? |
18:25.49 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B67BC5.pool.t-online.hu) |
18:25.50 | *** part/#asterisk lung (n=lung@24-148-96-186.ip.mhcable.com) |
18:26.23 | *** join/#asterisk juice (n=juice@mo-71-50-22-215.dhcp.embarqhsd.net) |
18:26.52 | *** join/#asterisk Kernel_core (i=Kernel_C@217.218.80.214) |
18:29.09 | *** join/#asterisk sasaim (n=root@202.5.145.13) |
18:29.47 | *** join/#asterisk g__ (n=g@itd01fw-fibre.itdepartment.com) |
18:30.47 | andrejkw | Is there any Phone Administration Menu available for Asterisk/ |
18:30.59 | *** join/#asterisk pdtmobile (n=ptinsley@209.12.249.243) |
18:30.59 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
18:31.09 | clive- | msg hi365 hi |
18:31.14 | hi365 | HI! |
18:31.26 | jbroome | hai2U |
18:32.41 | rob0 | heat index of 103F (39C) so far, forecast to go up to 108F (42C) ... whew |
18:32.46 | *** join/#asterisk pdtmobile (n=ptinsley@209.12.249.243) |
18:32.56 | rob0 | and I am thinking of going out into that heat |
18:33.29 | rob0 | car AC isn't working and the engine itself is overheating. |
18:37.04 | *** part/#asterisk apocn (n=apo@225stb46.codetel.net.do) |
18:37.56 | [TK]D-Fender | andrejkw : vim, emacs, mc, gedit, kedit, and amillion others, take your pick. |
18:38.03 | tessier_ | What do people recommend these days for interfacing asterisk with a T-1 that does not involve a PCI card? |
18:38.15 | pdtmobile | are there any linux supported USB sound cards? |
18:38.21 | andrejkw | lol |
18:38.24 | andrejkw | I mean an IVR. |
18:38.26 | Nugget | ha ha ha |
18:38.27 | andrejkw | a pre-made one |
18:38.45 | pdtmobile | I want to add sound to a couple of PBXs without cracking the case |
18:39.35 | [TK]D-Fender | anderiv : IVR's are damn quick to make.... nothing to sweat over |
18:40.00 | E-bola | the recoding is the sucky part of ivr's |
18:40.03 | E-bola | setup is easy |
18:41.34 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
18:41.40 | TripleFFFF | whats lall this "Forcing Marker bit, because SSRC has changed" ? |
18:41.58 | gandhijee | tessier_: the only other option AFAIK is a phonebrige/redphone |
18:42.01 | gandhijee | whatever its called. |
18:42.05 | gandhijee | but that too has a PCI card |
18:42.16 | gandhijee | i guess it might be possible to use some cisco crap |
18:42.29 | tessier_ | gandhijee: I'm thinking Cisco, Mediatrix, Rhino, something like that. |
18:43.00 | jbalcomb | [TK]D-Fender: did you test the info i sent you? |
18:43.03 | gandhijee | why not use a PCI card? |
18:43.19 | tessier_ | gandhijee: No echo cancelling, no signal processing, every PC is different. |
18:43.41 | gandhijee | umm, there is echo cancelling on the cards |
18:43.43 | gandhijee | just more $$ |
18:44.04 | [TK]D-Fender | jbalcomb : Yes, haven't had a chance to set my system up for it yet though. Also I'm a little unclear on the network map but I'll ask about that later. |
18:44.06 | [TK]D-Fender | jbalcomb : PM |
18:44.16 | gandhijee | you could always dedicate a PC as a PSTN bridge and use the heavy echo cancellers on them |
18:44.18 | tessier_ | I actually had decent luck with my digium PCI card but their POTS cards were a disaster and the support sucked. |
18:44.39 | tessier_ | So I am reluctant to go with them again. |
18:44.54 | tessier_ | Actually, the support on getting the PRI card going was great. |
18:45.00 | gandhijee | i haven't has any problems with the Digium TDM400 |
18:45.03 | tessier_ | The support on fixing all of my broken POTS cards was what was terrible. |
18:45.07 | gandhijee | there is always the option of sangoma. |
18:45.19 | tessier_ | Yeah, I have been looking at Sangoma. |
18:45.38 | [TK]D-Fender | tessier_ : they're GOLD |
18:46.14 | [TK]D-Fender | tessier_ : *0* echo with their DSP and no PCI issues. |
18:46.14 | gandhijee | i have them for for a PRI card, and use a digium for the POTS line |
18:46.41 | gandhijee | i think i've managed to port zaptel to ARM/Xscale i think |
18:46.55 | tessier_ | [TK]D-Fender: You are saying that Sangoma is gold? |
18:47.02 | hi365 | there really are good. if only i could get mine to work! |
18:47.05 | gandhijee | yea, sangoma is very good |
18:47.14 | [TK]D-Fender | tessier_ : As is great quality and solid like a rock. |
18:47.21 | gandhijee | hi365: haven't you been trying to get them for like a week now |
18:47.23 | tessier_ | [TK]D-Fender: What does it have over the digium PRI card? |
18:47.25 | g__ | Sangoma is very good for digital interfaces. |
18:47.34 | clyrrad1 | This is my first time to upgrade asterisk - what is the proper way to do it? I am on digium ftp and see a few options for the 1.2.10 including some "PATCH" files - to do the upgrade do I just download the new source and rebuild asterisk? |
18:47.35 | g__ | I have some reservations on their analogue products. |
18:47.36 | gandhijee | hi365: i told you that you should look at their wiki |
18:47.51 | hi365 | gandhijee feels like a year. it worked till i tried to add more channels |
18:47.58 | gandhijee | yeah Sangoma made the original T1/E1 cards |
18:48.00 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
18:48.01 | [TK]D-Fender | tessier_ : 3.3V/5V interoperable, high quality EC onboard, Doesn't mind sharing interrupts |
18:48.09 | [TK]D-Fender | tessier_ : Fors starters |
18:48.16 | hi365 | benn there done that more times then i remember |
18:48.49 | clyrrad1 | TKD- Have you upgraded asterisk before? |
18:49.17 | [TK]D-Fender | clyrrad1 : Plenty of times. |
18:49.22 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
18:49.25 | clyrrad1 | am I correct in my above statment? |
18:49.40 | clyrrad1 | to just download the new source and rebuild? |
18:49.53 | *** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju) |
18:50.03 | gandhijee | g__: why not fix them yourself, and help those guys out |
18:50.07 | [TK]D-Fender | clyrrad1 : download all the new and COMPLETE source files from FTP, wipe your old extracted source. Wipe your modules folder. Recompile everything. DONE |
18:50.46 | clyrrad1 | Excellent thanks :) |
18:51.09 | *** join/#asterisk AlexCTI (n=alex@adsl-074-238-025-003.sip.mia.bellsouth.net) |
18:51.14 | [TK]D-Fender | g__ : Funny that SUCCESS can be found in its entirely in the word SOURCE ;) |
18:51.15 | g__ | gandhijee: I'm a sysadmin, so I have no free time.. |
18:51.37 | g__ | The thought had crossed my mind though. |
18:51.42 | *** join/#asterisk johnnyb (n=jonathan@adsl-38-9-196.tulsaconnect.com) |
18:51.52 | gandhijee | O |
18:52.03 | *** join/#asterisk auralia_aNew (n=none@207.71.51.162) |
18:52.33 | g__ | TKD: of course, just because it's open source, doesn't mean it's gold. |
18:53.07 | [TK]D-Fender | g__ : I wasn't referring to open source so much as "compile the damn thing yourself you lazy bastard" ;) |
18:53.48 | g__ | TKD: I have too many servers for that.. pre-packaging has saved my ass a couple of times allready. |
18:53.58 | [TK]D-Fender | g__ : and its the fact is a solid product that makes it "gold". Closed products can earn that rating from me as well. I prefer "open" but it isn't the be-all and end-all. |
18:54.29 | gandhijee | g__: then you should be doing it yourself if you need the prepackaged |
18:55.30 | g__ | gandhijee: lovely idea. Sangoma's build script is a bit warpped. I started working on this and I asked them for help.. and they responded with "oh, the script can build debian packages itself".. |
18:56.01 | gandhijee | i wouldnt know about debian |
18:56.02 | _MDC_ | where can I found a good explanaition (and definition of) FXO, FXS, ISDN, digital/analog PSTN? |
18:56.04 | gandhijee | i use gentoo on my stuff |
18:56.17 | gandhijee | _MDC_: voip-info.org probably |
18:56.27 | g__ | Debian is just like gentoo.. except the patching and compiling happens ahead of time. |
18:56.30 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
18:56.41 | g__ | (doesn't know much about Gentoo) |
18:56.54 | gandhijee | i used debian and slack for about a whole 2 days for devel |
18:57.02 | gandhijee | then i trashed it all and went back to gentoo |
18:57.16 | g__ | reminds me of my experience with XP. |
18:57.24 | gandhijee | i never could get my cross-devel stuff to work right with debian |
18:57.25 | [TK]D-Fender | _MDC_ : Here..... |
18:57.26 | [TK]D-Fender | ~book |
18:57.28 | jbot | from memory, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
18:57.33 | gandhijee | so i just did it by hand on gentoo |
18:57.50 | g__ | In the end, it's all the same software.. I don't really care where it comes from. |
18:57.54 | gandhijee | true |
18:58.13 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
18:58.31 | _MDC_ | [TK]D-Fender, i'm reading the book right now, but just wanted a good explanation table... will read further |
18:59.02 | [TK]D-Fender | _MDC_ : I've read up on it in the book and the quality of the background info was pretty good and thorough |
18:59.19 | _MDC_ | ok, thanks, will read on |
19:00.18 | Dovid | my polycom wont prevision. |
19:00.22 | g__ | Question for everyone: I'm having occasional one-way audio problems over phone calls from PRI <-> IAX2 <-> SIP. (The callee can't hear the caller and eventually hangs up.) What troubleshooting steps can I take? |
19:00.37 | Dovid | i change the directory.cfg file but the phone wont grab it, any ideas ? |
19:00.55 | g__ | (There is no IAX2 trunking; debugging isn't enabled, but that could be done;.. ) |
19:01.12 | g__ | Dovid: check the ftp server logs |
19:01.21 | g__ | See what files it *is* grabbing. |
19:01.35 | Dovid | i did |
19:01.39 | g__ | And? |
19:01.42 | Dovid | says there wre no changes |
19:01.45 | Dovid | there wre |
19:01.59 | g__ | the ftp server log says that? |
19:02.08 | g__ | What files does it download? |
19:02.15 | *** join/#asterisk apocn (n=Apo@225stb46.codetel.net.do) |
19:02.19 | Dovid | none |
19:02.31 | g__ | What ftp server are you using? |
19:02.43 | Dovid | vsftpd |
19:02.47 | g__ | good! |
19:02.47 | Dovid | it was working b4 |
19:02.50 | *** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
19:02.54 | apocn | Is it possible to originate a call from one extension to another from the CLI? |
19:03.04 | Dovid | apocn: yes |
19:03.09 | g__ | How does it know which ftp server to access? DHCP information? |
19:03.16 | Dovid | no |
19:03.19 | Dovid | i put it in |
19:03.22 | Dovid | not using dgcp |
19:03.24 | Dovid | dhcp* |
19:03.37 | Dovid | i went and got clean configs from digium |
19:03.49 | Dovid | and it should wipe it clean but it isnt |
19:03.49 | g__ | And it's not even downloading 0000xxxxx.cfg? |
19:04.03 | Dovid | using the default |
19:04.13 | apocn | Dovid: where can I read about this? |
19:04.26 | Dovid | should i make it its own cfg file ? |
19:04.33 | Dovid | apocn: voip-info.org |
19:04.41 | g__ | Definately! if you have more than one phone, this is a must. |
19:04.57 | apocn | does it have a particular name? |
19:05.12 | Dovid | http://pastebin.ca/91586 |
19:05.18 | Dovid | using only one phone now |
19:05.26 | clyrrad1 | TKD - Upgrade worked thanks bro :) |
19:05.34 | Dovid | apocn: u wana make ur own app ur do it from the CLI ? |
19:05.47 | apocn | yeah |
19:05.57 | apocn | I made a webapp similar to FOP |
19:06.05 | g__ | You should see download attempts of 000f4xxxx.cfg, 000000000.cfg (if the former doesn't exist), bootrom.ld, 000f4xxxx-phone.cfg, 000f4xxxx-directory.cfg, 0000000000-directory.cfg (if the former doesn't exist).. etc |
19:06.23 | Dovid | can u look at my pastebin ? |
19:06.24 | apocn | using Ajax and PHP. Right now you can see queues/agents, and you can move an agent from one queue to another |
19:06.27 | Dovid | http://pastebin.ca/91586 |
19:06.34 | jbalcomb | Dovid: it can get the ftp server name from dhcp |
19:06.47 | Dovid | i dont have dhcp now |
19:06.49 | [TK]D-Fender | Dovid : You are modifying 00000000-directory.xml for your phone to use? |
19:06.50 | jbalcomb | Dovid: did you create the PlcmSpIp user? |
19:07.11 | [TK]D-Fender | Dovid : you can hardcode the provisiong sever right on the phone. |
19:07.31 | Dovid | jbalcomb: no,i redid the phone and put in diff ftp log in into |
19:07.33 | [TK]D-Fender | jbalcomb : And don't you tell me that you LEFT it at default! |
19:07.52 | Dovid | TK: i am tryin to get paging working, from what i understand I have to edit the fiels and then have the phone grab it |
19:08.14 | jbalcomb | Dovid: my polycom phones provision right out of the box. i copy <mac>.cfg and phone<exten>.cfg and they work. |
19:08.33 | Dovid | what do u mean by copy ? |
19:08.46 | apocn | Dovid, but now I want that when the supervisor clicks on the agent, communicate them both (like calling from his softphone to the agent) |
19:08.53 | jbalcomb | [TK]D-Fender: youre damn right i did. i'll fix it as i go. |
19:09.04 | jbalcomb | ;) |
19:09.05 | g__ | Dovid: your phone's logs are interesting, but your vsftpd logs would be more helpful. Mine show up in /var/log/auth.log |
19:09.09 | *** join/#asterisk NDT (n=nunya@cpe-24-195-66-214.nycap.res.rr.com) |
19:09.15 | Dovid | apocn: dont know the commands by heart. type in help and it will give em to u |
19:09.33 | apocn | ok, thanks |
19:09.53 | rene- | can i unpause an agent from the cli?? |
19:10.04 | Dovid | i dont have /var/log/auth.log |
19:10.40 | [TK]D-Fender | jbalcomb : :O |
19:10.41 | Dovid | g__: what do u mean that they are interesting ? |
19:10.50 | NDT | my dial command goes to a macro...when I run an agi script from inside the macro I am setting a variable in my agi script...when the call is completed and I return to after dial and go to h exten...I can't retrieve the contents of teh variable...so what happens to it? LOL Assuming this has something to do with setting it while I am in the macro |
19:11.23 | Dovid | TK: did u look at my pb ? |
19:11.43 | gandhijee | anyone here use and of the TDMoE stuff? |
19:11.45 | [TK]D-Fender | Dovid : Yes, and it says nothing about picking up its directory files. |
19:12.00 | gandhijee | i get a TDMoX:no master, is that normal? |
19:12.18 | Dovid | hmm |
19:12.20 | Dovid | y would that be ? |
19:12.43 | [TK]D-Fender | Dovid : What is the name of the directory file you modified and the phone doesn't see as different? |
19:12.48 | clyrrad1 | TDK - I found the MWI problem :) :) |
19:12.59 | [TK]D-Fender | clyrrad1 : And it was... ? |
19:13.00 | Dovid | also if i wipe out all files in my ftp directory and copy all the files fresh from the polycom zip shouldnt the phone revert to out of the box ? |
19:13.11 | [TK]D-Fender | Dovid : NO. |
19:13.23 | clyrrad1 | TKD- notifymimetype=text/plain |
19:13.25 | [TK]D-Fender | Dovid : Answer my last question please... |
19:13.27 | clyrrad1 | I commented that out and it works |
19:13.36 | [TK]D-Fender | clyrrad1 : Where was that occuring? |
19:13.40 | clyrrad1 | sip.conf |
19:13.46 | Dovid | 000000000000-directory.xml |
19:13.51 | clyrrad1 | under [general] |
19:14.16 | [TK]D-Fender | Dovid : That file WILL NOT WORK. only on the FIRST boot up of a new phone.... |
19:14.21 | Dovid | the manual said i dont need to set it as <mac>-directory.xml if i am only using one phone |
19:14.26 | Dovid | oh ok |
19:14.31 | clyrrad1 | how crazy - that one line was driving me nuts LOL |
19:14.34 | *** join/#asterisk eliel (n=eliel@200.123.183.89) |
19:14.44 | Dovid | so if i wana prevision i have to use <mac>-file.cfg ? |
19:14.57 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
19:15.24 | *** join/#asterisk _4d4m_ (n=adam@62.69.102.99) |
19:15.27 | NDT | Let me rephrase that last question....If you set a variable in AGI "SET VARIABLE blahblah" but you are inside a macro when you call the AGI script...what happens to the variable when you come out of the macro? I can't retrieve it when it returns to the h exten in the contect the macro was called from |
19:15.28 | *** join/#asterisk X-Rob_ (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
19:15.54 | NDT | context even |
19:17.24 | Dovid | TK: just fid <mac>.cfg and it still wont grab it |
19:17.50 | Dovid | one sec |
19:18.08 | rob0 | Maybe a dumb question, sorry, but what happens if I SetCallerID on a call going out a residential PSTN line? Does the telco's Caller ID cancel it out? (This is BellSouth, if that matters.) |
19:18.53 | rob0 | Kind of hard to test because I have no cell phone and I don't pay BellSouth for caller ID service. |
19:19.39 | [TK]D-Fender | Dovid : pastebin this file you just mentioned (including its exact name).... |
19:19.50 | NDT | ahh nevermind it is destroying that channel variable at hangup... |
19:19.56 | syzygyBSD | rob0: you have to have the ability to set the callerid, normal pots lines don't let you set it |
19:21.52 | rob0 | Yeah, this is just a home telephone line, no extra services at all. |
19:22.09 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-138-106.red.bezeqint.net) |
19:22.36 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
19:22.36 | Dovid | TK: one sec, phone |
19:23.05 | rene- | rob0: apparently setoutgoing wiil only work on digital lines and not in every case |
19:23.12 | rob0 | Phone?!? You mean people here actually TALK on those things? Ewwww. |
19:23.12 | rene- | setoutgoing callerid that is |
19:23.27 | tzafrir_laptop | we had to move to a different room at work, and suddenly we have a severe echo problem |
19:24.04 | tzafrir_laptop | (not from the phone or anything. The achustics of the room itself. Go figure...) |
19:24.08 | rob0 | tzafrir_laptop: put a big cork board on the wall :) |
19:24.47 | Dovid | TK: http://pastebin.ca/91607 |
19:25.19 | trelane_ | anyone here that uses snom phones mind sending me a couple of their stock configs for mass deployment? |
19:25.24 | trelane_ | or publishing them somewhere |
19:26.33 | g__ | tzafrir_laptop: I recommend putting carpet scraps on the wall. |
19:26.50 | g__ | ..Or put cubical dividers around. |
19:27.08 | g__ | Plants, bookshelves.. |
19:27.13 | rob0 | Crucify a boss. Nail him to the wall. |
19:27.22 | g__ | Yup--people also work. |
19:27.39 | rob0 | Believe it or not, they absorb a lot of sound while they're decomposing. |
19:27.51 | stoffell_h | 2 will even work better... |
19:27.54 | g__ | I didn't know that. |
19:28.06 | rob0 | 'Course the smell might get bad after a bit, but not as bad as while they were alive. |
19:28.10 | Dovid | TK: Also alook here http://pastebin.ca/91612 |
19:28.31 | Dovid | TK: and the other is http://pastebin.ca/91607 |
19:28.48 | g__ | Pictures of your boss? |
19:29.00 | trelane_ | tzafrir_laptop, http://froogle.google.com/froogle?q=acoustic+foam&hl=en&btnG=Search |
19:29.23 | g__ | expensive, eh? |
19:29.44 | trelane_ | g__: yeah acoustic foam is not cheap |
19:29.50 | Dovid | TK: u there ? |
19:29.57 | trelane_ | you don't need much of this stuff, get the heavy foam and do only one wall and it should kill the whole room |
19:30.26 | trelane_ | g__, you ought to see how much the adhesive costs |
19:30.34 | g__ | How much is one room's worth? |
19:30.40 | trelane_ | g__, what room size? |
19:30.53 | g__ | this room. |
19:31.02 | trelane_ | this is a channel not a room, if you want a chat room try AOL People Chat? |
19:31.06 | g__ | how about a small bedroom. |
19:31.12 | trelane_ | say 10x14? |
19:31.18 | g__ | Sure. |
19:31.33 | stoffell_h | g__: depends on what country you are in ... some small bedrooms can be pretty big.. |
19:31.59 | g__ | That's true.. North American standards. |
19:32.26 | g__ | (We'll catch up with the rest of the world eventually.) |
19:32.36 | stoffell_h | ;) |
19:32.51 | stoffell_h | that's not always the best thing to do. lol |
19:33.06 | Dovid | where r my polycom people ? |
19:33.34 | jbroome | they ran away, along with your "a" and "e" |
19:33.47 | stoffell_h | and they were fast.. |
19:34.03 | gandhijee | anyone here use TDMoE? |
19:34.08 | gandhijee | i am apparently having some problems with it |
19:34.54 | Dovid | [TK]D-Fender: please let me know when u return |
19:35.07 | trelane_ | g__: the Sonomatt stuff is about $950 for the foam, assume another 50-100 for the adhesive for said room, it breaks down to roughly $18/linear foot with 8' ceilings |
19:35.19 | jbroome | oh god, the "y" and "o" ran away too! |
19:35.49 | g__ | Wow.. I'm going to stick with carpet scraps and towels myself, thanks. |
19:36.20 | trelane_ | g__, right but with this stuff you really only SHOULD do one wall, I priced all 4 (As that's how I do recording studios which must be totally dead (high pile carpet and sonomatt'd ceiling as well |
19:36.32 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
19:36.32 | *** join/#asterisk Hmmhesays (n=Neg@24-117-135-28.cpe.cableone.net) |
19:36.53 | Zodiacal | can differnt brands of phones and even different protocals view/update hints? |
19:37.01 | Zodiacal | i.e. cisco sccp with a polycom sip? |
19:37.27 | Zodiacal | the hints are managed by asterisk right so it should really mater. or am i dreaming? |
19:37.42 | trelane_ | g__, consider what it'd cost to carpet all 4 walls in a room |
19:37.46 | Zodiacal | should = shoudn't |
19:38.05 | g__ | trelane: if their scraps, it would be a labour-only cust. |
19:38.09 | g__ | cost, rather. |
19:38.21 | Zodiacal | why carpet a wall? |
19:38.24 | trelane_ | g__, where do you know of to get free carpet scraps that's sufficient to do this? |
19:38.30 | trelane_ | Zodiacal, acoustics |
19:38.38 | trelane_ | Zodiacal, carpet kills echo |
19:38.43 | Zodiacal | trelane i know someone that used egg cartens |
19:38.48 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
19:38.49 | Zodiacal | for a band |
19:38.59 | Zodiacal | nailed them into plywood and hung that |
19:39.00 | g__ | trelane: any carpet place throws out tons of the stuff during installation. I'm sure if you ask around you could get enough. |
19:39.02 | Zodiacal | worked ok |
19:39.38 | eKo1 | carpet scraps + plywood == sound proof wall? |
19:40.00 | Zodiacal | egg cartens + plywood |
19:40.03 | Zodiacal | theres somthing about the shape |
19:40.04 | g__ | Zodiacal: I've heard that one too.. I've also heard towels.. and someone recommended hanging a Duvet on a line. |
19:40.05 | trelane_ | eKo1, not sound proof, echo canceling |
19:40.19 | eKo1 | echo canceling? |
19:40.31 | trelane_ | eKo1, I'm talking, it echos off the wall, makes the room fscking loud |
19:40.50 | trelane_ | downsides: carpet retains heat, carpet looks odd on walls |
19:40.54 | Hmmhesays | mattresses |
19:40.59 | trelane_ | Hmmhesays, indeed |
19:41.08 | Hmmhesays | no joke |
19:41.21 | [TK]D-Fender | Dovid : thsoe files don't tell me anything bout the XML contact directory you are working on. |
19:41.23 | Zodiacal | any ideas about my 'hint' question? |
19:41.39 | *** join/#asterisk m4rkl4r (n=markp@outboundemail.uneta.com) |
19:42.14 | trelane_ | g__, one more note, carpet on the walls depending on zoning and fire code may not be legal, consult a construction/archetectural type before doing it |
19:42.20 | stoffell_h | Zodiacal: u want to know if polycom's and cisco's work well in 1 setup ? (even with hints) |
19:42.26 | Dovid | TK: so what do u need ? |
19:42.31 | Zodiacal | stoffell_h yep |
19:42.31 | Dovid | what files ? |
19:44.03 | Dovid | [TK]D-Fender: wut u need ? |
19:44.04 | stoffell_h | Zodiacal: it'll work.. but... i wouldn't do it... |
19:44.41 | Zodiacal | why not |
19:45.06 | Zodiacal | they will rarly talk with each other |
19:45.14 | Zodiacal | mostly to transfer calls etc.. |
19:45.27 | [TK]D-Fender | Dovid : You need to make your directory file per-phone as <mac>-directory.xml |
19:45.30 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B67BC5.pool.t-online.hu) |
19:45.33 | Dovid | i am doing that |
19:45.49 | stoffell_h | Zodiacal: if I want to 'manage' 100 phones.. i prefer them to be 100 phones of brand X, not 30 phones brand X and 70 brand Y.. just for simplicity.. and manageability... |
19:46.00 | Zodiacal | ic |
19:46.30 | stoffell_h | Zodiacal: standardizing is, in this case, a good thing to do.. :) |
19:46.30 | trelane_ | Zodiacal, pick one, or the other, don't do both. You will regret it later |
19:46.39 | [TK]D-Fender | Zodiacal : Yes, * maintains status based on ITS awareness. Meaning if you DIRECTLY call a SIP phone its "monitoring" then * will not know the call is in progress. |
19:47.09 | Zodiacal | directly call? |
19:47.10 | [TK]D-Fender | Zodiacal : Anything * is responsible for it propagates status for as well as the channel driver permits (can't speak for SCCP, only SIP). |
19:47.22 | Dovid | files are called 0004f2052338-directory.xml |
19:47.30 | [TK]D-Fender | Zodiacal : To which I can say that an IP 601 + 3 Attendent modules = really cool..... |
19:47.33 | Dovid | and 0004f2052338.cfg |
19:47.40 | [TK]D-Fender | Dovid : then it should pick them up. |
19:47.47 | Dovid | it isnt, can u think of y ? |
19:47.52 | [TK]D-Fender | Dovid : pastebin 0004f2052338-directory.xml |
19:48.05 | stoffell_h | [TK]D-Fender: lol, i just got my 430 (since a few days), can't wait to start playin' with it |
19:48.42 | [TK]D-Fender | stoffell_h : I got mine last week.... yes, VERY cool, and the firmware is MCUH zippy-er than any of the others, and boots much faster as well. |
19:49.05 | [TK]D-Fender | stoffell_h : I like the LCD skin they use and wish they ported its look&feel to the others. |
19:49.11 | Dovid | TK: http://pastebin.ca/91639 |
19:49.16 | [TK]D-Fender | stoffell_h : Feels very CCS-ish |
19:49.45 | [TK]D-Fender | Dovid : Looks OK, guess you should verify the user rights on it. |
19:49.48 | eKo1 | What does this message actually mean: Got SIP response 488 "Not Acceptable Here" back from 192.168.53.29 |
19:50.02 | stoffell_h | [TK]D-Fender: nice! i just had the time to power it up for 10secs, the blinkin' lights are cool, gotta look into the rest.. rrrr ;) |
19:50.05 | eKo1 | Wh is it not acceptable? |
19:50.10 | Dovid | TK: do the files have to be set as read only ? |
19:50.55 | [TK]D-Fender | eKo1 : Codec mis-match |
19:51.18 | [TK]D-Fender | Dovid : I suggest 644 to the user. |
19:51.28 | Dovid | hmm |
19:51.31 | Dovid | now it just worked |
19:51.33 | Dovid | ok |
19:51.45 | Dovid | this time i did a hard reboot |
19:51.47 | Dovid | oh well |
19:51.51 | Dovid | looks like its workin |
19:51.52 | Dovid | thanks |
19:51.52 | [TK]D-Fender | Dovid : you DO have to reboot to take changes naturally. |
19:52.03 | Dovid | reboot ? |
19:52.21 | Dovid | turn on phone, get new files, turn off phone and then it will have it ? |
19:52.33 | Dovid | or turn on get files and then there ? |
19:52.49 | Dovid | i did hard reboot as oposed to just rebooting from phone menu |
19:52.52 | [TK]D-Fender | turn off phone, mod the file, turn on phone, done |
19:53.04 | Dovid | ok |
19:53.22 | Dovid | let me try again, just to be sure its workin, thanks for all the time |
19:53.44 | eKo1 | [TK]D-Fender: right. thanks |
19:54.17 | file | [TK]D-Fender: how are we today? |
19:54.21 | [TK]D-Fender | I should get a job as paid remote Polycom support :) |
19:54.39 | [TK]D-Fender | file : we are well.... |
19:54.46 | Dovid | hehe |
19:54.48 | Dovid | yes u should |
19:55.04 | [TK]D-Fender | I mean I pimp them enough you'd THINK I was paid to.... |
19:55.10 | Dovid | lol |
19:55.16 | file | admit it - you are! |
19:55.19 | Dovid | i learn from people like , |
19:55.21 | jbalcomb | dovid: you might need to reset the local config on the phone as well |
19:55.38 | Dovid | i pass around the luv. was in here 2 hours workin wit some one, givin em help |
19:55.43 | [TK]D-Fender | jbalcomb : not for phone directory. if it doesn't match it trusts the server. |
19:55.48 | Dovid | jbalcomb: what do u mean ? |
19:55.58 | [TK]D-Fender | jbalcomb : not like a config override requires. |
19:56.05 | jbalcomb | [TK]D-Fender ah, ok. thanks polycom guy. |
19:56.20 | [TK]D-Fender | "pseudo"-Polycom guy! |
19:56.25 | stoffell_h | ;) |
19:56.31 | stoffell_h | pseudo ? :) |
19:56.38 | jbalcomb | [TK]D-Fender how many shares of thier stock do you own? |
19:56.39 | Dovid | now its workin :):):):) |
19:56.43 | Dovid | lol |
19:56.48 | [TK]D-Fender | jbalcomb : none :( |
19:56.58 | jbalcomb | [TK]D-Fender LIAR!! |
19:57.00 | Dovid | ok, now that i have that, anyone wana help with polycom paging ? |
19:57.09 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
19:57.10 | [TK]D-Fender | jbalcomb : HEY! There's something for you to make! Polycom Buddly list editor! |
19:57.22 | jbalcomb | [TK]D-Fender maybe that'll be next |
19:57.27 | [TK]D-Fender | jbalcomb : That is worth GUI-ing |
19:57.37 | jbalcomb | [TK]D-Fender everything i do is worth doing pal. |
19:57.51 | [TK]D-Fender | jbalcomb : As long as you're being PAID to ;) |
19:58.15 | *** join/#asterisk Druken (i=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com) |
19:58.46 | Druken | ok... i feel dumb, but am i wrong in thinking that nat=yes is correct? |
19:58.49 | syzygyBSD | that is the "worth" part |
19:58.54 | *** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net) |
19:59.01 | syzygyBSD | depends on if you have a nat... |
19:59.12 | Druken | syntax.... |
19:59.27 | syzygyBSD | I have seen that in the tutorials, but I prefer nat=1 |
19:59.34 | [TK]D-Fender | Druken : Depends on the question ;) |
19:59.53 | Druken | [TK]D-Fender: i have nat=yes in my sip.conf |
20:00.05 | [TK]D-Fender | Druken : I believ it accepts either, but ask yourself this... what does NAT=3 mean? :) |
20:00.08 | Druken | yet.. the damn sip show peers isn't showing it as nat |
20:00.23 | [TK]D-Fender | Druken : pastebin.... |
20:01.36 | [TK]D-Fender | I need an IP 601 to keep up with all the customers I'm maintaining! |
20:02.29 | Dovid | i luv the 601 |
20:02.29 | Druken | http://pastebin.ca/91651 |
20:02.33 | Dovid | i jus need to do paging |
20:03.35 | [TK]D-Fender | Druken : WIP (Unspecified) D N 255.255.255.255 0 UNKNOWN |
20:03.49 | Druken | correct |
20:03.56 | [TK]D-Fender | Druken : I might think that it would not fill in that field becuase perhaps the phone isn't registered.... |
20:04.09 | [TK]D-Fender | Druken : Reg it and see what happens |
20:04.26 | Druken | well of course it's not registered... cause it's behind the nat and needs the flag set |
20:04.42 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
20:04.51 | Druken | damn phone has been trying to register for hours now... |
20:05.12 | *** join/#asterisk af_ (n=af@ip-170-209.sn1.eutelia.it) |
20:05.30 | [TK]D-Fender | Druken : Do other phones of yours work behind NAT currently? |
20:05.50 | Druken | dunno, don't have any at the moment... |
20:06.01 | [TK]D-Fender | Druken : pastebin your [general] section. |
20:06.23 | Druken | it's only 4 lines... |
20:06.27 | *** join/#asterisk iCEBrkr (i=icebrkr@69.9.167.70) |
20:06.38 | *** join/#asterisk [Airwolf] (n=airwolf@dsl51B67CB3.pool.t-online.hu) |
20:06.56 | Druken | http://pastebin.ca/91657 |
20:08.59 | [TK]D-Fender | Druken : Does your * server have a public IP? |
20:09.33 | Druken | uhmm... yes.. i'm not a complette moron :) |
20:10.16 | stoffell_h | hm, what's the capital of canada? |
20:10.21 | *** join/#asterisk Unistim_junky (n=rover@c-71-56-28-13.hsd1.ga.comcast.net) |
20:10.23 | mitcheloc | ottowa! |
20:10.27 | [TK]D-Fender | Druken : on sip debug do you see the incoming registration attempt? |
20:10.28 | Druken | uhmm... toronto? :) |
20:10.35 | mitcheloc | quebec! |
20:10.44 | Druken | on sip debug i see so much shit, who can read it? |
20:10.48 | stoffell_h | lol, 3 diferent answers... :) |
20:10.49 | [TK]D-Fender | mitcheloc : Actually the first capital of Canada was MONTREAL <- |
20:10.54 | mitcheloc | better yet, canada has a government? |
20:11.03 | jbroome | syrupistan |
20:11.04 | stoffell_h | i'm from europe, sorry i don't know... |
20:11.21 | [TK]D-Fender | stoffell_h : Ottawa, Ontario |
20:11.41 | stoffell_h | [TK]D-Fender: thanks, a reliable source ;) (google confirms) |
20:11.42 | [TK]D-Fender | Druken : no excuses! |
20:12.01 | mitcheloc | ooh, was i right with ottawa? |
20:12.02 | Druken | pfft... i think it's a great excuse... hehe |
20:12.27 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
20:12.28 | mitcheloc | Druken, i agree, good point, nobody *can* read it, yet everyone points to it... |
20:12.41 | [TK]D-Fender | Druken : BTW some routers screw the hell out of SIP regardless and you may be up a creek. What kind of phone? |
20:12.47 | mitcheloc | you need special elite speed reading skillZ |
20:13.00 | Druken | WIP300, behind an rt31p2 |
20:13.02 | [TK]D-Fender | mitcheloc : I do OK on sip debug, and I'm far from experienced compared to many here |
20:13.17 | *** part/#asterisk TESTER2 (n=Cyber@modemcable082.42-81-70.mc.videotron.ca) |
20:13.24 | Unistim_junky | Where can I see a list of commands I can do by calling asterisk with -rx? |
20:13.30 | [TK]D-Fender | Druken : OH!!!! WIP300 doesn't work behind NAT! its a serious POS. forget the router... its the PHONE. |
20:13.50 | Dr-Linux | i'd agree SIP seems nothing, but it's a huge game |
20:13.50 | Druken | hmm.. well it worked last night... |
20:13.56 | [TK]D-Fender | SIP WiFi = SUCK |
20:14.15 | mitcheloc | sip is a virus |
20:14.20 | [TK]D-Fender | Druken : from behind NAT? I've read up on their lack of NAT functionailty. |
20:14.30 | mitcheloc | it's like a weed that grew up in the voip space and should be stamped out |
20:14.36 | Druken | that's the only way it could have run in my house... |
20:14.42 | [TK]D-Fender | mitcheloc : Life is a sexually transmitted disease which is in all cases FATAL. |
20:14.45 | Druken | i don't have a block of ip's here... |
20:14.55 | stoffell_h | lol mitcheloc |
20:14.56 | [TK]D-Fender | Druken : Hrm.... COULD be the router then... |
20:14.57 | *** join/#asterisk AJaymn (i=AJmn@70.59.126.197) |
20:14.59 | mitcheloc | [TK]D-Fender, thanks for ruining my day fender |
20:15.19 | Druken | maybe i'll just setup a server for my house... |
20:15.28 | Druken | fucken pain in the ass |
20:17.00 | Dr-Linux | lol |
20:17.01 | [TK]D-Fender | PITA.... good for sandwichs, bad for IT.... |
20:17.10 | eKo1 | hehehe |
20:18.25 | jbalcomb | feh, i now have several phones that only have one way audio. |
20:18.38 | Unistim_junky | can a user be unregistered via the CLI? |
20:18.48 | jbalcomb | they can hear the person they call but the person cant here them. also, no one can call them. |
20:18.49 | Assid | [TK]D-Fender: how do you know which routers act stupid with nat? |
20:19.03 | [TK]D-Fender | Assid : Trial & Frustration. |
20:19.14 | Hmmhesays | ok this as5300 is still giving me hell |
20:19.14 | AJaymn | Anyone know of a VoIP provider that allows CID info (spoofing) ??? |
20:19.17 | Dr-Linux | jbalcomb: codec's issue? |
20:19.28 | Dr-Linux | jbalcomb: what CLI says? |
20:19.30 | [TK]D-Fender | AJaymn : Apparently Broadvoice.... |
20:19.35 | jbalcomb | Dr-Linux: i'll double check but everything is ulaw here |
20:19.55 | Unistim_junky | jbalcomb : check your firewall |
20:20.02 | Dr-Linux | jbalcomb: what's softphone/hardphones users have? |
20:20.12 | [TK]D-Fender | jbalcomb : Would all of the defect phones happen to carry the name "Grandstream"? |
20:20.20 | jbalcomb | Unistim_junky: no firewall internal and it just started happening about two hours ago. |
20:20.40 | jbalcomb | [TK]D-Fender: nope, one x-ten lite and one polycrap ip501 |
20:20.43 | Unistim_junky | jbalcomb: could be routing. |
20:20.49 | [TK]D-Fender | jbalcomb ;) |
20:20.51 | AJaymn | [tk]d-fender ive never been able to get it to work with them |
20:20.53 | jbalcomb | [TK]D-Fender ;) |
20:20.57 | [TK]D-Fender | jbalcomb : All local LAN? |
20:21.02 | jbalcomb | [TK]D-Fender certainly |
20:21.16 | [TK]D-Fender | jbalcomb : Hrm |
20:21.24 | jbalcomb | [TK]D-Fender i thought at first there was something about dtmfmode... |
20:21.36 | [TK]D-Fender | jbalcomb : No, that should all be OOB anyways... |
20:21.54 | jbalcomb | [TK]D-Fender i swicthed a gxp-2000 with a polycom and had to update the sip.conf.. maybe thats when it starts.. with the 'sip reload' |
20:21.59 | [TK]D-Fender | jbalcomb : 1 way typically runs to NAT... or perhaps a defective handset / headset |
20:22.00 | Dr-Linux | [TK]D-Fender: what a call listen, when call to those users? |
20:22.24 | Hmmhesays | Anyone good at cisco as5300's? |
20:22.32 | [TK]D-Fender | Dr-Linux : Talks does funny Yoda... hhmmmMMMMMMM?!?!?! |
20:22.46 | jbalcomb | [TK]D-Fender: hey, maybe this is the first reload i've done since you had me change those settings? |
20:23.13 | [TK]D-Fender | jbalcomb : pastebin your general section for me, and list the IP's of the devices involved. |
20:23.13 | Dr-Linux | jbalcomb: wht asterisk version? |
20:23.29 | jbalcomb | Dr-Linux 1..5 |
20:23.45 | [TK]D-Fender | 1.5?! OMG, he's ahead of us all!!! |
20:23.53 | [TK]D-Fender | Even DIGIUM! |
20:24.00 | Dr-Linux | lol |
20:24.13 | Dr-Linux | he means 1.2.5 |
20:24.18 | Dr-Linux | i believe :P |
20:24.41 | jbalcomb | Dr-Linux: i developed my own source tree and moved on.. ;) |
20:24.52 | [TK]D-Fender | jbalcomb : Probably more like POISON OAK ;) |
20:25.07 | Dr-Linux | jbalcomb: then you maybe have alot of other problems as well :P |
20:25.18 | jbalcomb | haha.. yeah, perhaps my terrible code could be viewed as such |
20:25.40 | Dr-Linux | jbalcomb: did you try restarting your asterisk? |
20:26.04 | [TK]D-Fender | jbalcomb : pastebin qwuick before I'm outta here... |
20:26.05 | jbalcomb | [TK]D-Fender AH HA!! the receptionist PC running x-ten lite is on 192.168 |
20:26.18 | [TK]D-Fender | jbalcomb : you can add multiple |
20:26.23 | [TK]D-Fender | "localnet" clauses.... |
20:26.26 | jbalcomb | [TK]D-Fender http://pastebin.ca/91685 |
20:26.36 | Dr-Linux | jbalcomb: there could be many reasons i.e NAT, Firewall, Qaulify, Codec uncompatibility ... and so |
20:27.11 | [TK]D-Fender | jbalcomb : http://pastebin.ca/91688 |
20:27.19 | rob0 | Well, I have * set up as a softphone now ... works, but it's still a softphone :) |
20:27.21 | [TK]D-Fender | jbalcomb : ywc |
20:27.34 | jbalcomb | ywc? |
20:27.40 | [TK]D-Fender | You're WelCome |
20:27.48 | jbalcomb | ah, ty |
20:27.58 | *** join/#asterisk TommyTheKid (n=tommythe@mpk-edge.cto.sunit.net) |
20:28.01 | gandhijee | anyone know why when i run my ztcfg i only get 25 out of my 30 channels for TDMoE??? |
20:28.05 | rob0 | Really laggy even calling a Zaptel phone on the same Ethernet segment. |
20:28.26 | Dr-Linux | gandhijee: ztcfg -vvv |
20:28.27 | rob0 | like about 1 second lag times. |
20:28.55 | gandhijee | i know |
20:28.59 | Dr-Linux | gandhijee: otherwise check your zaptel.conf |
20:29.03 | gandhijee | it only configures 25 lines |
20:29.06 | gandhijee | its standard. |
20:29.14 | TommyTheKid | Does anyone have any advice on dual core opterons verses dual physical CPUs? does it matter? .. plan to use a quad PRI (maybe 2) in it as a conferencing server |
20:29.15 | gandhijee | only diff is its running on Xscale/IXP |
20:29.28 | gandhijee | but that shouldn't matter for that driver, it doesn't interface to the PCI bus |
20:29.33 | Dr-Linux | gandhijee: what about your zaptel.conf? |
20:30.10 | Unistim_junky | can a user be registered and unregisted via the Asterisk CLI> |
20:30.28 | [TK]D-Fender | ok, heading home, BBIAB |
20:30.31 | Dovid | TK: having problems with the wiki and on paging |
20:30.44 | Dr-Linux | jbalcomb: you have only ulaw |
20:30.47 | Dr-Linux | add some more |
20:30.57 | jbalcomb | Dr-Linux all the phones are ulaw |
20:31.00 | Dovid | what do i need to edit in sip.conf to have the phone accept paging ? |
20:31.09 | Unistim_junky | balcomb: If its all internal why do you have an externalip setup |
20:31.13 | Dr-Linux | ilbc takes lot of CPU and it's slim, that's why skype uses |
20:31.14 | Dr-Linux | :) |
20:31.41 | Unistim_junky | Dr-Linux: can a user be registered and unregisted via the Asterisk CLI> |
20:31.42 | TommyTheKid | so, do we have chan_skype.so yet? :) |
20:31.43 | Dr-Linux | jbalcomb: that would be not a pain if you try more .. |
20:32.05 | *** part/#asterisk JohnJacob (n=dhorner@pool-71-127-102-43.aubnin.fios.verizon.net) |
20:32.24 | gandhijee | nm, i figured it out |
20:32.37 | Dr-Linux | Unistim_junky: hhm... does user's account axist? |
20:32.47 | Unistim_junky | yes |
20:33.02 | Unistim_junky | so they are in sip.conf |
20:33.28 | Unistim_junky | so I want to be able to unregister / register via CLI |
20:34.01 | gandhijee | i had it as 2-31 |
20:34.14 | gandhijee | i forgot that the TDM400 takes 1 to 4 regardless |
20:34.28 | Unistim_junky | Dr-Linux: I figure if I can do it on CLI> then I can right a cron job to do it. |
20:35.08 | gandhijee | i dont need to load asterisk on the side that is exporting my TDMoE spans do it? |
20:35.10 | gandhijee | *i |
20:35.21 | Dr-Linux | hhm.. |
20:35.32 | Dr-Linux | Unistim_junky: i'm trying but today my browsing sucks |
20:35.42 | Dr-Linux | Unistim_junky: did you try google? |
20:35.50 | tzafrir_laptop | TommyTheKid, don't count on such a chan_skype any time soon |
20:36.04 | TommyTheKid | hehe |
20:36.05 | tzafrir_laptop | TommyTheKid, BTW: have you tried google talk? |
20:36.20 | TommyTheKid | nope, as far as I know, its d0ze only |
20:36.27 | Unistim_junky | Dr-Linux: google, voip-info, digium... I figured I must be searching wrong keywords |
20:36.44 | tzafrir_laptop | On linux there are some other clients |
20:37.06 | tzafrir_laptop | For chat alone you can use any jabber client |
20:37.43 | TommyTheKid | my real question was regarding Dual Core CPUs, like the Opteron 180, does dual core vs dual physical CPU make a difference with asterisk/digium te412p? Also would it be better or worse to do 64 bit? |
20:38.15 | TommyTheKid | yea, I use Psi or ichat on my googleTalk account, but its not voice :) |
20:39.12 | tzafrir_laptop | dual core is basically dual CPU. |
20:39.30 | tzafrir_laptop | (not to be mixed withhh "twice as fast") |
20:39.39 | TommyTheKid | that was my view on it.. its not like HT :) |
20:39.47 | *** join/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net) |
20:39.54 | tzafrir_laptop | TommyTheKid, latest development psi should support using Jingle |
20:40.02 | tzafrir_laptop | Haven't tried it |
20:40.11 | clive- | i was wondering that also, does a dual core 3 ghz compare with a xeon 2 x 3 ghz ? |
20:40.33 | TommyTheKid | I fon |
20:40.35 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
20:40.59 | TommyTheKid | uh, I don't use Intel, but I think the Xeon CPUs tend to be a bit more "server class" (higher cache, etc) than the CoreDUO? |
20:41.38 | *** join/#asterisk VoIPMasta (n=John@201.160.17.205.cableonline.com.mx) |
20:41.47 | gandhijee | there is a CoreDUO based Xeon now |
20:41.57 | TommyTheKid | ah |
20:42.09 | lokkju | well, the Core Duo is coming out in multiple versions - a mobile, a workstation, and a server class chip, if I remmember right |
20:42.10 | gandhijee | we just devel'd a platform for EMC using that arch |
20:42.10 | VoIPMasta | Hi there, I'm setting an IVR but for an unknown reason it's not detecting the numbers I dial to reach the extensions, any ideas? |
20:42.20 | *** join/#asterisk tsurk0 (n=tsurko@85.187.160.157) |
20:42.47 | rob0 | I must say though, I feel considerably geekier with a * installed on /dev/laptop. Highly recommended to anyone who feels the need for increased geekitude. |
20:43.05 | Strom_C | rob0: I have two laptops with asterisk installed |
20:43.08 | Strom_C | i win |
20:43.22 | rob0 | yes you do ... I wasn't trying to compete with you :) |
20:43.26 | gandhijee | i think i have managed to port zaptel to Xscale for this board we have at this place |
20:43.32 | VoIPMasta | rob0: I 've had asterisk running on my laptop for at least 1 year |
20:43.49 | gandhijee | board has x86 and IXP/Xscale on the same PCB sharing the PCI bus |
20:43.54 | rob0 | Now let's persuade Digium to make CardBus Zaptel adapters. |
20:44.01 | Unistim_junky | rob0: I have asterisk running on a 3850 IPAQ |
20:44.23 | TommyTheKid | Digium specific Q: The te412p (we are testing) has quad T1/E1 capability with DSP/echo canceling/etc... so that must mean it can handle 30 line E1's which means that we have like 12 (actually 16) DSPs that are not in use, can we use those for other uses? :) |
20:44.24 | Unistim_junky | I think I win |
20:44.28 | VoIPMasta | Unistim_junky: I have also asterisk running on an embedded WRAP |
20:45.16 | Unistim_junky | nice, what is WRAP |
20:45.29 | VoIPMasta | have you seen SOEKRIS embedded mother boards? |
20:45.37 | rob0 | wireless router appliance? |
20:45.38 | Unistim_junky | o... ok |
20:45.41 | Unistim_junky | sorry.. |
20:45.45 | VoIPMasta | something like that |
20:46.00 | VoIPMasta | a small Geode 266 Mhz CPU, with a 256MB CF card |
20:46.06 | gandhijee | VoIPMasta: IXP465 gonna test out the TDM interfaces soon |
20:46.19 | VoIPMasta | it does also have 128 Mb RAM and 2 100mbps NICs |
20:46.37 | Unistim_junky | Do you guys have any comments on registering / unregistering a user via CLI |
20:47.06 | *** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net) |
20:47.12 | TripleFFFF | whats most stable version of asterisk ? |
20:47.12 | VoIPMasta | and does anyone have any ideas on why my IVR isn't detecting the digits I dial to access extensions? |
20:47.21 | VoIPMasta | TripleFFFF: go for the latest |
20:47.23 | TripleFFFF | that could be prodction wised |
20:47.28 | jbalcomb | VoIPMasta: perhaps a dtmfmode problem? |
20:47.29 | TripleFFFF | well im getting weird shit |
20:47.44 | Unistim_junky | dtmf... |
20:47.56 | VoIPMasta | I have 1.2.9.1 in a production environment |
20:48.01 | VoIPMasta | and I haven't had any problems with it |
20:48.17 | *** join/#asterisk piwi (n=piwi@80-43-91-224.dynamic.dsl.as9105.com) |
20:48.18 | Unistim_junky | have you changed the loc. your calling from. |
20:48.20 | VoIPMasta | Unistim_junky: Yup, I think it;s something related to dtmf modes or something |
20:48.24 | piwi | hi there |
20:48.33 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
20:48.43 | VoIPMasta | Unistim_junky: yes, I've tried from my cell phone and from a SIP device using a regular analog phone |
20:48.50 | Unistim_junky | Depending on the where you call from they may have a different SBC |
20:49.09 | Unistim_junky | try changing dtmf to auto ....I think it is... |
20:49.25 | piwi | hey, is anyone using asterisk with freephonie.net? |
20:49.44 | VoIPMasta | however I'm setting up this IVR using a SIP-provisioned DID |
20:49.48 | Unistim_junky | a different gateway... |
20:49.53 | VoIPMasta | so I don't have a "context" in sip.conf for this DID |
20:49.58 | VoIPMasta | just a register line |
20:50.10 | jbalcomb | how does one reboot the cisco7940 from the keypad? |
20:50.17 | Dr-Linux | how many bucks a voip guy get paid for an hour in USA? |
20:50.27 | Unistim_junky | * + 6 + settings |
20:50.38 | VoIPMasta | Dr-Linux: depends on the tasks and knowledge |
20:50.44 | bkw_ | you guys want a laugh |
20:50.44 | bkw_ | http://crazytelemarketer.ytmnd.com/ |
20:51.05 | Dr-Linux | VoIPMasta: normally? |
20:51.41 | VoIPMasta | Dr-Linux: paid by the hour (something like a freelancer) or a signed contract? |
20:52.00 | *** join/#asterisk ariel_ (n=ariel_@74.8.35.2) |
20:52.04 | VoIPMasta | Unistim_junky: where can I set up the dtmf mode considering that I'm having this DID forwarded by a DID provider? |
20:52.21 | Dr-Linux | freelancer |
20:52.57 | Unistim_junky | general section of sip.conf |
20:52.57 | Dr-Linux | VoIPMasta: also tell me contract |
20:54.06 | Unistim_junky | VoIPMasta: What do you mean being forwarded? |
20:54.16 | VoIPMasta | Unistim_junky: it worked :) |
20:54.22 | Unistim_junky | The user is hitting your IVR right? |
20:54.33 | znoG | do many people use call parking? |
20:54.33 | Unistim_junky | VoIPMasta: Cool |
20:54.55 | Unistim_junky | Now, If Someone would just help me........ |
20:55.17 | Dr-Linux | VoIPMasta:? |
20:55.18 | NDT | Is the uniqueid...call specific or channel specific? |
20:56.09 | Unistim_junky | user meaning user in sip.conf |
20:56.11 | VoIPMasta | Dr-Linux: from what I know: freelancer ~30.00 / hour. |
20:57.14 | *** part/#asterisk TommyTheKid (n=tommythe@mpk-edge.cto.sunit.net) |
20:57.24 | VoIPMasta | Unistim_junky: I haven't registered a user using the CLI, so I'm afraid I'm unable to help you |
20:57.36 | Unistim_junky | NoProb.. me neither |
20:57.52 | Dr-Linux | VoIPMasta: aww someone ofered me $12/hr |
20:57.56 | VoIPMasta | Unistim_junky: to be honest, I didn't think it could be done, but it also depends on which protocol you're using |
20:58.09 | VoIPMasta | Dr-Linux: but what's your expertise/experience? |
20:58.31 | *** join/#asterisk fulgas (n=fulgas@a81-84-116-1.cpe.netcabo.pt) |
20:58.35 | daysmen3_ | HELP anyone see a problem with syntax here - exten => s,n,Set(EXTNUM=${IF($["${EXT:-1}" = "#"]?${EXT:1}:${EXT})}) |
20:59.18 | Dr-Linux | VoIPMasta: they need me for asterisk and hardphones setup .. |
20:59.25 | Unistim_junky | [TK]D-Fender: What do you think? Can a sip user be registered via the CLI |
20:59.31 | VoIPMasta | Dr-Linux: and how experienced are you at it? |
20:59.32 | Unistim_junky | setVar |
20:59.52 | Unistim_junky | daysmen3_: |
20:59.55 | *** part/#asterisk fulgas (n=fulgas@a81-84-116-1.cpe.netcabo.pt) |
21:00.17 | Dr-Linux | VoIPMasta: i'm already running 4 asterisk servers located in USA |
21:00.44 | ariel_ | wow 4 servers. |
21:00.44 | Dr-Linux | VoIPMasta: 2 years with network field and 8 months with asterisk |
21:00.57 | Hmmhesays | I need as5300 help |
21:00.58 | Hmmhesays | argh |
21:01.20 | Dr-Linux | VoIPMasta: they want me, bcoz i'm in/from Pakistan. |
21:01.29 | Unistim_junky | VoIPMasta: You a Cisco man? |
21:01.42 | VoIPMasta | Unistim_junky: I've played with cisco for some time |
21:01.46 | [TK]D-Fender | Unistim_junky : Not AFAIK |
21:01.50 | daysmen3_ | im getting a - expecting $end; """ = "#" |
21:01.58 | ariel_ | Hmmhesays, what is the issue with the unit? |
21:02.02 | VoIPMasta | Dr-Linux: Well, but you're overseas, that also means a lower wage |
21:02.21 | Dr-Linux | VoIPMasta: overseas? |
21:02.23 | VoIPMasta | Dr-Linux: a lot of companies in the US look for people to work from home (and from another country) because it's cheaper |
21:02.33 | *** join/#asterisk ivanfm (n=ivanfm@201.52.129.236) |
21:02.39 | rob0 | US$30/hour sounds low for professional * consulting. I charge much more for general Linux work. |
21:02.40 | Dr-Linux | VoIPMasta: correct |
21:02.56 | Dr-Linux | VoIPMasta: i'm already working for a US company |
21:03.06 | VoIPMasta | rob0: I do also charge more for a lot of things, but a "voip guy" like Dr-linux described gets about 30/hr |
21:03.10 | [TK]D-Fender | Cisco certs get $100/h on consulting. |
21:03.20 | ariel_ | not many here in the US charge less then 50 per hour. Most are over 100.00 |
21:03.30 | [TK]D-Fender | I typically charge $40/h |
21:03.43 | ariel_ | [TK]D-Fender, wow your cheap. |
21:03.44 | VoIPMasta | ariel_: there's a diff between a pro and an amateur consultant |
21:04.06 | VoIPMasta | there are literally thousands of geek kids that have been messing with asterisk for some months and offer "consulting services" |
21:04.08 | rob0 | What's the difference? The amateurs know more? :) |
21:04.09 | Dr-Linux | VoIPMasta: actually i'd be very happy if i get even $10/hrs in pakistan |
21:04.30 | VoIPMasta | rob0: sometimes the certificate grants you the right to charge more |
21:04.31 | ariel_ | VoIPMasta, hummm wonder if they people think that there only amateur. |
21:04.46 | VoIPMasta | rob0: like [TK]D-Fender said, a cisco certified consultant charges over 100/hr |
21:05.05 | VoIPMasta | that's why certifications are so useful |
21:05.11 | *** part/#asterisk sevard (n=sev@adsl-71-129-115-244.dsl.irvnca.pacbell.net) |
21:05.14 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
21:05.35 | VoIPMasta | Dr-Linux: don't worry, I'm from Mexico and wages here are also in the low side. |
21:05.39 | [TK]D-Fender | ariel_ : But not "easy" ;) |
21:05.53 | ariel_ | [TK]D-Fender, I see. |
21:05.55 | Dr-Linux | VoIPMasta: they also want me to configure webserver/iptables and trouble ticket system for them on linux |
21:06.09 | Dr-Linux | VoIPMasta: hehe cool |
21:06.15 | VoIPMasta | Dr-Linux: then charge them for ticket/event instead of charging by the hour |
21:06.15 | ariel_ | VoIPMasta, I see, Mexico deals more in Peso then dollars... |
21:06.30 | VoIPMasta | ariel_: not really, most of my rates are in dollars |
21:06.59 | Dr-Linux | i see |
21:06.59 | Dr-Linux | VoIPMasta: what you do? |
21:06.59 | VoIPMasta | ariel_: on IT industries most rates are in US |
21:06.59 | VoIPMasta | ariel_: on IT industries most rates are in USD |
21:07.08 | lesouvage | "show queues" output is : queue1 has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), W:0, C:1, A:0, SL:100.0% within 0s What does the Sl:100% within Os means? |
21:07.17 | VoIPMasta | Dr-Linux: I run 4 datacenters, plus manage some voip devices (gateways) and also do some coding |
21:07.26 | Unistim_junky | service level |
21:07.32 | Dr-Linux | VoIPMasta: good good |
21:07.34 | AJaymn | Anyone know of a VoIP provider that allows CID info (spoofing) ??? |
21:07.37 | Unistim_junky | lesouvage: |
21:07.53 | ariel_ | AJaymn, voipjet does |
21:07.53 | VoIPMasta | AJaymn: there are some in Europe that allow you to spoof it |
21:08.01 | Unistim_junky | Ajaymn: telasip & voicepulse |
21:08.04 | VoIPMasta | ariel_: but sometimes voipjet doesn't even pass on CID |
21:08.15 | Strom_C | hello |
21:08.24 | *** join/#asterisk pbx1 (n=pbx1@58.69.92.3) |
21:08.26 | ariel_ | VoIPMasta, yes if you send them the name. If you just send them the numbers there fine with it. |
21:08.28 | Dr-Linux | VoIPMasta: my later plans are outsourcing .. as i just started to make my a few sites i.e www.syednetworks.com , redhat.pk , tele.pk etc |
21:08.55 | AJaymn | Any with Unlimited US calling? |
21:08.59 | Unistim_junky | slashdot |
21:09.04 | lesouvage | Unistim_junky: ? |
21:09.07 | *** join/#asterisk Dibbler_ (n=Dibbler@dsl-217-155-254-174.zen.co.uk) |
21:09.19 | Unistim_junky | lesouvage: service level |
21:09.27 | rob0 | Unlimited? Limited only by how much you're willing to pay? |
21:09.58 | rob0 | or are you talking about a flat rate? |
21:10.07 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
21:10.18 | lesouvage | Unistim_junky: you mean 100 % of the incoming calls has been answered without delay? |
21:10.28 | rob0 | I think the per-minute plans are usually better. |
21:10.38 | rob0 | but I don't know |
21:11.43 | jbalcomb | Unistim_junky: thank you |
21:12.59 | rene- | are the grandstreams GSM capable? |
21:13.03 | rene- | that would be cool |
21:13.42 | Unistim_junky | jbalcomb: glad to help |
21:13.58 | Unistim_junky | lesouvage: Yeah, something like that |
21:14.47 | Unistim_junky | lesouvage: If I remember correctly you say all calls should be answered within like 5 minutes, and that pecentage shows how many were actually answered within that amount of time |
21:15.19 | Unistim_junky | so if a manager looks at it they can see whether or not the agents are jerking off or not |
21:15.33 | rene- | Voipmasta: do you work on your own or are you employed |
21:15.37 | rene- | I live in Mexico too |
21:15.55 | VoIPMasta | I am a shareholder of the company I work for |
21:16.01 | VoIPMasta | technically I'm the CEI |
21:16.02 | VoIPMasta | CEO |
21:16.09 | rene- | CTO? |
21:16.14 | rene- | ah |
21:16.16 | VoIPMasta | nope, CEO |
21:16.23 | rene- | where in mexico are you based? |
21:16.27 | TripleFFFF | man since when to PH sing ? |
21:16.27 | VoIPMasta | I have 51% of the shares |
21:16.39 | VoIPMasta | rene- We have offices in Mexico City, and Morelos |
21:16.40 | Azrael | VoIPMasta: is your company public? |
21:16.45 | rob0 | Whacha doing in IRC? You should be out making money? :) |
21:16.49 | rene- | hahah |
21:16.58 | carrar | man 1.2.9.1 is buggy |
21:16.59 | VoIPMasta | Azrael: by public do you mean shares offered in an open market? |
21:17.00 | rene- | he gets .01 per msg posted |
21:17.00 | TripleFFFF | irc is not CEO , only CTO 's ;) |
21:17.04 | Azrael | VoIPMasta: yes |
21:17.05 | Unistim_junky | is voip restricted in Mexico? |
21:17.17 | rene- | yes |
21:17.18 | TripleFFFF | yes mexico sucks for voip.. ask telmex and fox |
21:17.21 | VoIPMasta | Not really |
21:17.23 | VoIPMasta | it's not regulated |
21:17.24 | rene- | you can do it for your private use |
21:17.26 | VoIPMasta | telmex sucks |
21:17.27 | Azrael | VoIPMasta: publicly traded is what i mean |
21:17.28 | TripleFFFF | telmex is ran by fox grey alien friends |
21:17.32 | VoIPMasta | but there's not a real regulation for it |
21:17.38 | gandhijee | hey anyone know what package newt.h belongs too? |
21:17.38 | Strom_C | do any of you regularly purchase polycom phones, and if so, where do you purchase them from? |
21:17.44 | TripleFFFF | how real can a 9mm bullet be ? |
21:17.56 | VoIPMasta | TripleFFFF: nope, Telmex isn't run by Fox nor any of his friends, but by Carlos Slim (The richest man in Latin America) |
21:17.56 | Unistim_junky | YEs I heard that the goverment sends folks to chat to find out who is illegally routing calls outside of the place. |
21:18.06 | Azrael | Strom_C: at the last place i worked i setup the VoIP system for the company. i got the Polycom phones from voipsupply iirc |
21:18.07 | rene- | to say it simply, if they find out you have something as simple as a calling center doing voip calls you get busted |
21:18.14 | NDT | Why the heck does a jumping to a macro from dial command change the uniqueid of the call? |
21:18.19 | TripleFFFF | and how you think Slim to Fox relates to our candadian Desmarais and harper/bush ? |
21:18.20 | Azrael | Strom_C: actually that was just one of the vendors. i'd call a bunch out there and see if you can get quantity discounts. |
21:18.24 | VoIPMasta | rene-: we haven't been busted, and we've been doing voip for over 3 years |
21:18.32 | TripleFFFF | how can you be richest man without any politic connections ? |
21:18.35 | Strom_C | Azrael: I just want one for testing purposes |
21:18.37 | TripleFFFF | hence his friend |
21:18.38 | TripleFFFF | lol |
21:18.42 | Azrael | Strom_C: i worked with the Polycom 301 and 501 models - great phones |
21:18.50 | Azrael | Strom_C: you can probably call a vendor and setup a 30 day trial |
21:18.50 | rene- | voip for private usd is allowed |
21:18.55 | Unistim_junky | gandhijee: lib-newt |
21:18.59 | VoIPMasta | rene- VoIP for end users is allowed |
21:19.02 | Strom_C | Azrael: I've had a bad time trying to configure those phones |
21:19.06 | TripleFFFF | like bush declaring wars everywhere caus the company making the tanks etc weapons is his fathers, and borthers;lol |
21:19.09 | VoIPMasta | what is restricted is to provide VoIP services |
21:19.12 | rene- | not if terminates in PSTN |
21:19.13 | TripleFFFF | anyhow back to voip |
21:19.20 | rene- | ok |
21:19.28 | rene- | you can set up vonage for some friend |
21:19.29 | Strom_C | Azrael: so I figure I should buy one and learn how to do it properly |
21:19.31 | Azrael | Strom_C: yeah? i set them up for automagical TFTP config, wrote the xml config files, and it all worked great |
21:19.34 | VoIPMasta | rene- We have a company in the US, and we provide VoIP services "from" the US |
21:19.39 | TripleFFFF | im wondering if i should get BE edition of asterisk |
21:19.44 | VoIPMasta | rene- our company in Mexico just sales devices ;) |
21:19.45 | [TK]D-Fender | Strom_C : My rates are very accessable ;) |
21:19.54 | TripleFFFF | and if its 1k per server or per company lol |
21:20.04 | gandhijee | Unistim_junky: is that perl related stuff too? |
21:20.21 | Strom_C | Azrael: would you mind shooting over some sample xml configs? all the ones I've run into have been so horribly formatted that they've been impossible to work with |
21:20.24 | Unistim_junky | gandhijee: what dow you mean? |
21:20.27 | rene- | VoipMasta: i think you walk a thin line |
21:20.46 | Azrael | Strom_C: private msg me your email. i'll be able to get to it tonight. |
21:20.53 | gandhijee | well everything i search for on that returns stuff saying those are perl binds to redhats newt library |
21:20.59 | VoIPMasta | rene-: I have a "value added services" license from COFETEL as well as some pretty good lawyers |
21:21.03 | gandhijee | which i am assuming is newlib |
21:21.12 | rene- | value added service is not for voip |
21:21.15 | Strom_C | [TK]D-Fender: who do you like to buy polycom phones from? |
21:21.18 | Unistim_junky | nah I don't think so. |
21:21.31 | gandhijee | i've tried to compile and install just newlib (as i need it for an embedded system) but it doesn't seem to work |
21:21.31 | Dr-Linux | how someone can be Asterisk Certified ? |
21:21.32 | rene- | value added service is like $500 USD from COFETEL |
21:21.41 | gandhijee | DCAP |
21:21.45 | *** join/#asterisk teknoprep (n=chris@unaffiliated/teknoprep) |
21:21.48 | rene- | by taking DCAP exam and get over 70 in both the written and live test? |
21:21.50 | VoIPMasta | rene-: but it's enough to provide IVR, voice messaging, and such services |
21:21.51 | Strom_C | I just got my dcAP cert today! |
21:21.52 | [TK]D-Fender | Strom_C : CCP (Canadian Communications Products) - www.ccpin.com |
21:21.56 | teknoprep | hey all |
21:22.09 | rene- | congrats Strom_C |
21:22.12 | teknoprep | with trixbox.. its basically redhat? |
21:22.13 | Dr-Linux | rene-: how much fees? |
21:22.22 | teknoprep | i can use it for other small apps if need be? |
21:22.25 | rene- | i only paid for the exam and that was like 200-300 usd |
21:22.25 | [TK]D-Fender | Strom_C : They are an auth'd reseller though I know more about them than they do ;) |
21:22.33 | Dr-Linux | rene-: can i take exams being in Pak? |
21:22.44 | rene- | i dont know abouit that Dr-Linux |
21:22.53 | rene- | ask oej the DCAP guy |
21:22.57 | Dr-Linux | rene-: are you asterisk certified? |
21:23.07 | rene- | yes |
21:23.12 | Strom_C | Dr-Linux: the exam has to be taken in person |
21:23.27 | Strom_C | Dr-Linux: there is a practical and a written portion |
21:23.50 | Dr-Linux | Strom_C: means i can't do? :( |
21:23.58 | Dr-Linux | i can't go to USA |
21:24.28 | Strom_C | Dr-Linux: i think there are tests scheduled at astricon europe |
21:24.33 | ariel_ | Dr-Linux, he is in the EU most of the time. |
21:24.37 | gandhijee | Dr-Linux: i belive its offered in europe somewhere |
21:24.53 | [TK]D-Fender | bkw_ : Thats some funny shit.... |
21:25.01 | Dr-Linux | aww |
21:25.14 | Dr-Linux | what's the info site for DCAP? |
21:25.33 | rene- | jbot: google |
21:25.34 | jbot | extra, extra, read all about it, google is a search engine found at http://www.google.com/ |
21:25.34 | ariel_ | ~ dcap |
21:25.36 | jbot | methinks dcap is Digium Certified Asterisk Professional. See http://www.voip-info.org/tiki-index.php?page=Asterisk+dCAP |
21:25.38 | Dr-Linux | actually i wanna try Asterisk certification :S |
21:25.57 | file | dCAPitated! |
21:29.09 | Dr-Linux | rene-: i can't do DCAP :( |
21:30.02 | piwi | hey guys I was wondering, I have asterisk running as a daemon on my machine, can I login from ekiga to the asterisk pbx from this same machine? |
21:30.56 | VoIPMasta | piwi: yes you should be able to |
21:31.09 | Dr-Linux | gandhijee: no problem maybe someday Digium know that there is a country pk, till then i'll learn some agi :) |
21:31.23 | VoIPMasta | piwi: just make sure that asterisk is listening for SIP (i assume you're using SIP in ekiga) on localhost (or your NIC IP) |
21:31.51 | gandhijee | lol |
21:31.54 | [TK]D-Fender | Dr-Linux : Try working on the basics first before asking to be certified. |
21:31.57 | [TK]D-Fender | ;) |
21:32.18 | *** join/#asterisk SkramX (n=MarkS@admins.sentiensystems.net) |
21:32.20 | Strom_C | Dr-Linux: it's a difficult test |
21:32.23 | gandhijee | Dr-Linux: why can't you go to Europe to take it? |
21:32.26 | Strom_C | hi SkramX |
21:32.37 | droops | hey stron |
21:32.38 | SkramX | How would I specify a variable for an AGI script in a file that I put in var/spool/asterisk/outgoing/ ? |
21:32.41 | VoIPMasta | I can give you a AUFPUOC certificate for free |
21:32.42 | piwi | VoIPMasta: which config file do I have to go in to change that? |
21:32.44 | Strom_C | hi droops |
21:32.46 | SkramX | Hiya, Strom_C |
21:32.49 | SkramX | Heya, droops too |
21:32.53 | droops | hey SkramX |
21:32.56 | SkramX | we know eachother via that BinRev place, eh? |
21:32.57 | droops | long time on see |
21:32.57 | Dr-Linux | [TK]D-Fender: well, i just wanted to try :) |
21:33.03 | droops | on = no |
21:33.17 | Strom_C | haha yep |
21:33.18 | SkramX | droops: Yeapps |
21:33.22 | SkramX | going to HOPE? |
21:33.23 | Strom_C | that good ol binrev thing |
21:33.24 | VoIPMasta | piwi: sip.conf |
21:33.25 | Dr-Linux | gandhijee: can't go to Europe/USA/CA .. Visa and shit you know |
21:33.29 | droops | not me |
21:33.43 | SkramX | I think Strom_C already said he wasn't either |
21:33.45 | Strom_C | Dr-Linux: guess what - I've got my dCAP cert now :) |
21:33.49 | SkramX | anyone wanna help me out though? :) |
21:33.49 | Strom_C | er |
21:33.52 | Strom_C | droops: |
21:33.54 | piwi | VoIPMasta: cheers! Im sorry Im total newbie with asterisk, starting from scratch |
21:34.00 | droops | good deal Strom_C |
21:34.01 | *** join/#asterisk TommyTheKid (n=tommythe@mpk-edge.cto.sunit.net) |
21:34.06 | Strom_C | SkramX: whats teh problem? |
21:34.14 | SkramX | How would I specify a variable for an AGI script in a file that I put in var/spool/asterisk/outgoing/ |
21:34.17 | *** join/#asterisk teknoprep (n=chris@unaffiliated/teknoprep) |
21:34.22 | droops | did strom just misspell something |
21:34.24 | VoIPMasta | piwi: great, keep up the good work |
21:34.46 | TommyTheKid | How many quad PRI (te411p or 412p) cards can I put in say a quad 2.4GHz Opteron? are there sizing guidelines anywhere? |
21:34.53 | SkramX | i have a php script that places a call by putting the .call file in the proper /var directory.. but I want to make a string for Cepstral/an agi script to recite.. |
21:34.56 | AJaymn | Anyone using ShellShark for VOIP ???? |
21:34.56 | Dr-Linux | Strom_C: well, actually certification doesn't matter in my country as very rare guys know here "what is Asterisk" |
21:35.08 | Strom_C | Dr-Linux: that comment was supposed to be for droops |
21:35.11 | gandhijee | Tommy: i with the digium stuff a max of 2 |
21:35.12 | Strom_C | Dr-Linux: damned autocomplete |
21:35.19 | gandhijee | i mean i heard |
21:35.24 | TommyTheKid | ok |
21:35.33 | Strom_C | TommyTheKid: I've seen more than two cards in a single box |
21:35.38 | gandhijee | sangoma makes an 8 port T1/E1 card |
21:35.43 | Strom_C | TommyTheKid: but that can get hairy |
21:35.46 | *** join/#asterisk jm|work (n=jamiem@dsl-217-155-242-137.zen.co.uk) |
21:35.46 | bkw_ | yes they do |
21:35.51 | SkramX | heh |
21:36.02 | SkramX | thanks bkw_ ! |
21:36.07 | Strom_C | SkramX: I asked you to tell me the problem :) |
21:36.08 | droops | i dont know SkramX, but im waiting for the answer |
21:36.10 | TommyTheKid | I plan to use it for conferencing too, so how many say 15-20 person conferences can ride onm the same box |
21:36.11 | jm|work | helo :) |
21:36.23 | SkramX | Strom_C: well, I need to pass the .call file some sort of variable |
21:36.24 | TommyTheKid | I assume with the onboard DSP cards, there isn't much load on the CPU just for the call itself |
21:36.26 | jm|work | I have Googled for this but I seem to be going in circles: |
21:36.29 | jm|work | app_dial.c: Unable to create channel of type 'Zap' (cause 0 - Unknown) |
21:36.31 | [TK]D-Fender | bkw_ : Basically the punchline was the *slick* at the end? |
21:36.33 | jm|work | :/ |
21:36.33 | SkramX | I dont know have a specific error or something |
21:36.40 | Dr-Linux | Strom_C: well, whenever i face any problem with any of my asterisk box i ask here, even i solved the problem.. but i wanna see other peoples veiws .. |
21:36.43 | gandhijee | Tommy: like i said sangoma makes an 8 port T1/E1 card |
21:36.55 | VoIPMasta | jm|work: what card are you using? |
21:36.59 | Dr-Linux | Strom_C: and sometime i get very good help |
21:37.10 | jm|work | wcfxo0: <Wildcard X100P> port 0xb000-0xb0ff mem 0xee030000-0xee030fff irq 16 at device 8.0 on pci0 |
21:37.12 | SkramX | bkw_: nice |
21:37.13 | TommyTheKid | gandhijee: I saw that, how many sangoma cards can we put in? |
21:37.16 | jm|work | Found a Wildcard FXO: Wildcard X100P |
21:37.16 | jm|work | ZapTel device loaded. |
21:37.16 | jm|work | Registered tone zone 4 (United Kingdom) |
21:37.18 | SkramX | i recently got a macmini core duo |
21:37.20 | TommyTheKid | we are looking for density here :) |
21:37.27 | Strom_C | jm|work: please use pastebin |
21:37.30 | Strom_C | ~pb |
21:37.34 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/ |
21:37.34 | VoIPMasta | jm|work: is it a digium x100p or a "clone"? |
21:37.35 | bkw_ | TommyTheKid, sangoma cards do not have IRQ issues |
21:37.43 | bkw_ | so you can put many in the same box |
21:37.44 | jm|work | VoIPMasta: no sure ..... |
21:37.47 | jm|work | not* |
21:37.56 | VoIPMasta | jm|work: where did you buy it? |
21:38.02 | jm|work | recently |
21:38.02 | TommyTheKid | with quad CPU's I seriously doubt I would have IRQ issues anyhow :) |
21:38.05 | [TK]D-Fender | TommyTheKid : Not that you'd NEED to at 8ports / card :) |
21:38.09 | VoIPMasta | where, not when |
21:38.12 | jm|work | it was quite cheap and "off eBay" so I daresay it's a clone |
21:38.17 | *** join/#asterisk znoG (n=gs@162-148-235-201.fibertel.com.ar) |
21:38.19 | VoIPMasta | under 15? |
21:38.24 | jm|work | er |
21:38.25 | Dr-Linux | [TK]D-Fender: did you remember my asterisk box? you compiled 7 months ago? :) |
21:38.36 | VoIPMasta | how much did you pay for it? |
21:38.39 | bkw_ | if you come to cluecon you'll get a free T1 card :P |
21:38.41 | [TK]D-Fender | Dr-Linux : Yup (somewhat) |
21:38.46 | jm|work | "It's Real X100P FXO card for Digium Asterisk pbx OEM" according to the speil |
21:38.49 | jm|work | yeah <15 |
21:39.00 | bkw_ | its a freakin INtel 537 Modem |
21:39.04 | *** part/#asterisk Unistim_junky (n=rover@c-71-56-28-13.hsd1.ga.comcast.net) |
21:39.04 | Dr-Linux | [TK]D-Fender: heh that server uptime is 212 days :) |
21:39.04 | bkw_ | the same thing digium was selling for 99 bucks |
21:39.10 | VoIPMasta | I'm pretty sure it's a clone with one of the components busted so that it's detected as a "real X100p" |
21:39.18 | jm|work | :( |
21:39.22 | bkw_ | I really like how digium was selling 6 dollars modems to everyone for so long at 99 bucks a pop |
21:39.24 | *** join/#asterisk kcortez (n=kcortez@208.49.103.100) |
21:39.24 | VoIPMasta | jm|work: use pastebin and paste your zaptel.conf file |
21:39.35 | bkw_ | they pop two resistors off and slap on a heat sink |
21:39.37 | Dr-Linux | [TK]D-Fender: but now i have 4 servers in US datacenter |
21:39.44 | jm|work | it only has one line, really |
21:39.50 | mishehu | bkw_: I'm in the wrong business |
21:39.50 | jm|work | fxsks=1 |
21:39.52 | VoIPMasta | bkw_: When I found out that I started buying those modems wholesale |
21:39.59 | bkw_ | I did too |
21:40.02 | VoIPMasta | right now I have over 500 "clone" x100p cards |
21:40.06 | Dr-Linux | [TK]D-Fender: i never bother to upgrade 1.2.0 to latest one |
21:40.13 | NDT | For awhile Newegg had em for $5 |
21:40.20 | piwi | VoIPMasta: sorry but Im still not making it :( I put in general defaultip=localhost and when I launch ekiga it says port for sip is in use and it doesnt even try to login to asterisk and says it failed to register |
21:40.29 | bkw_ | VoIPMasta, but most people feel that if you're not buying hardware from digium you're morally in the wrong |
21:40.41 | VoIPMasta | piwi: tell ekiga to listen on a different port (other than 5060) |
21:40.47 | jm|work | VoIPMasta: am I screwed :( |
21:40.51 | bkw_ | like if you use Asterisk with say OpenVox cards... you're evil! |
21:40.57 | jm|work | "if it looks too good to be true: it probably is" |
21:41.10 | VoIPMasta | bkw_: I'm not selling them, I use them for my company |
21:41.27 | VoIPMasta | bkw_: and when I bough those modems it was almost impossible to buy digium hardware here in mexico |
21:41.29 | Dr-Linux | Strom_C: i confess i'm not good, but imagine my country. i got award for 2006 |
21:41.33 | VoIPMasta | so I take no guilt for using them |
21:41.44 | *** join/#asterisk saftsack (n=saftsack@p54A7E68B.dip.t-dialin.net) |
21:41.47 | bkw_ | I use sangoma hardware for all new projects |
21:41.54 | [TK]D-Fender | Dr-Linux : and you're PROUD of that?! Wak up to the "now". You should fix your setup to 1.2 spec and upgrade for security alone. |
21:42.04 | mishehu | I will be using sangoma for all future projects. |
21:42.13 | *** join/#asterisk lokkju (n=lokkju@unaffiliated/lokkju) |
21:42.27 | saftsack | bkw_, whats the reason for this decision? |
21:42.36 | bkw_ | at least with sangoma I can move to freeswitch once we have the functionality complete |
21:42.45 | VoIPMasta | jm|work: hold on |
21:42.46 | saftsack | what is freeswitch? |
21:42.50 | jm|work | VoIPMasta: okies thanks :) |
21:42.53 | bkw_ | saftsack, www.freeswitch.org |
21:42.55 | [TK]D-Fender | bkw_ : No zaptel channel driver? |
21:42.59 | SkramX | Strom_C: I think I found the solution... ill speak up later |
21:43.10 | Strom_C | SkramX: uh, ok |
21:43.11 | bkw_ | [TK]D-Fender, if you really knew how zaptel worked you would not be using it either |
21:43.13 | Corydon-w | saftsack: because bkw_ initially forked Asterisk, then moved to a completely rewritten project when that fork died |
21:43.20 | [TK]D-Fender | bkw_ : Or is it so far off the beaten track for your uniform structure to be "feasable"? |
21:43.23 | Cresl1n | pssh.... you're just bitter bkw_ |
21:43.28 | Cresl1n | you don't know how it works either |
21:43.28 | bkw_ | No i'm not |
21:43.30 | [TK]D-Fender | bkw_ : I'll take that as a "yes" |
21:43.35 | bkw_ | Cresl1n, yes I do |
21:43.39 | Dr-Linux | [TK]D-Fender: you guided me alot and i followed you, if you remember i asked you "i hve no problems with current version should i upgrade" you said ...new versions new problems... |
21:43.42 | Cresl1n | whateva foo |
21:43.48 | Cresl1n | not like I do |
21:43.50 | saftsack | Corydon-w, is this forked project openpbx? |
21:43.53 | jm|work | VoIPMasta: the guy that was selling it has sent me a zapata.conf |
21:44.00 | Corydon-w | saftsack: yeah, openpbx |
21:44.08 | jm|work | not sure if that helps |
21:44.16 | mishehu | openpbx died because it was too hard to maintain I bet. |
21:44.17 | [TK]D-Fender | Dr-Linux : That doesn't carry when there are KNOW full-on crash exploits and major versions that threaten maintainability. |
21:44.17 | piwi | VoIPMasta: cant change the port with ekiga, do you know any good voip client easy to install with ubuntu? |
21:44.24 | bkw_ | OpenPBX is still alive |
21:44.25 | *** join/#asterisk anthm (n=anthm@adsl-69-211-76-234.dsl.milwwi.ameritech.net) |
21:44.25 | *** mode/#asterisk [+o anthm] by ChanServ |
21:44.27 | Cresl1n | bkw_: you just talk about things that you don't know adequately about |
21:44.28 | mishehu | freeswitch is completely different. |
21:44.30 | VoIPMasta | piwi: linphone |
21:44.33 | [TK]D-Fender | OMGZ, teh man! |
21:44.34 | bkw_ | Cresl1n, not true |
21:44.35 | saftsack | Corydon-w, why did this fork died? |
21:44.55 | VoIPMasta | piwi: it's in the synaptic packet manager (if you add all the branches) |
21:45.01 | saftsack | or are there any news belong this diieng? |
21:45.01 | Corydon-w | saftsack: because not enough people thought it was worthwhile and the reasons that they forked were addressed |
21:45.02 | mishehu | bkw_: I thought they stopped trying to maintain it. |
21:45.08 | *** join/#asterisk MRH2 (n=Mr_happy@host-83-146-30-242.bulldogdsl.com) |
21:45.19 | VoIPMasta | jm|work: use a pastebin and paste your zapata.conf there |
21:45.25 | VoIPMasta | jm|work: not here, in a pastebin |
21:45.27 | [TK]D-Fender | bkw_ : Why fork * when you can just build FreeSwitch from the ground up right? |
21:45.28 | jm|work | ok ok! |
21:45.29 | jm|work | :) |
21:45.32 | saftsack | ok, thats not good because i thought openpbx was a good idea |
21:45.49 | bkw_ | [TK]D-Fender, and actually build it correctly |
21:45.56 | [TK]D-Fender | bkw_ : Hail Mary |
21:45.58 | Dr-Linux | [TK]D-Fender: ok, my other 3 servers have 1.2.9.1 but THIS one 1.2.0 i'll upgrade it on weekend |
21:46.14 | bkw_ | I just have differing views on how things should be done vs what Digium has on the roadmap for Asterisk |
21:46.15 | Corydon-w | saftsack: at its peak, it had maybe 5 developers, 4 of which have since left for greener pastures |
21:46.19 | mishehu | [TK]D-Fender: a house on a bad foundation will always sink into the ground. |
21:46.22 | VoIPMasta | I have some very old asterisk versions running on some servers |
21:46.26 | jm|work | VoIPMasta: http://pastebin.ca/91754 |
21:46.36 | VoIPMasta | but since those servers are on a lan I'm not really concerned about security issues |
21:46.52 | [TK]D-Fender | bkw_ : To me, as much as I love *, its greatest value to me is its REPLACABILITY. Thats why I'm all for Polycom / Sangoma. The "core" doesn't own me, and never will. |
21:46.56 | mishehu | doesn't matter if you try to move the house and keep the foundation. |
21:46.58 | saftsack | Corydon-w, thats not good :( |
21:47.05 | TommyTheKid | wow, sangoma has Solaris support.. hmmmmm :) |
21:47.06 | jm|work | VoIPMasta: and here is zaptel.conf http://pastebin.ca/91756 |
21:47.15 | Corydon-w | saftsack: and really, the problem with openpbx was that the prima donnas left the Asterisk project and tried to collaborate... and prima donnas can only collaborate for so long before they start to bicker amongst themselves |
21:47.15 | Dr-Linux | anybody ever luck to configure Sphinx voice recognition system with asterisk? |
21:47.16 | [TK]D-Fender | bkw_ : So when Freeswitch hits the threshold point we'll see where the dust settles. |
21:47.17 | bkw_ | [TK]D-Fender, I can agree with that one |
21:47.42 | bkw_ | [TK]D-Fender, I like options |
21:47.52 | VoIPMasta | jm|work: try running ztcfg -vv |
21:47.55 | MRH2 | TK - speaking of polycom do you happen to know how far off SIP2.0 is from a release date? |
21:48.03 | Corydon-w | saftsack: so basically, openpbx imploded when they couldn't agree on a path forward |
21:48.11 | [TK]D-Fender | bkw_ : Yup, standards first, control second, source third... |
21:48.14 | saftsack | Corydon-w, hmm ok sounds logical |
21:48.16 | jm|work | VoIPMasta: http://pastebin.ca/91757 |
21:48.29 | [TK]D-Fender | MRH2 : I haven't touched base with them lately. I'm runing 2.0 beta right now. |
21:48.37 | saftsack | Corydon-w, but the reasons why openpbx was forked were logical in my opinion |
21:48.45 | Corydon-w | saftsack: freeswitch is basically a project from the ground up by one of the former openpbx developers |
21:48.48 | MRH2 | aye ok |
21:48.54 | bkw_ | Corydon-w thats wrong |
21:49.04 | bkw_ | Tony was never a developer on OpenPBX he just gave advice |
21:49.08 | mishehu | Corydon-w: fud |
21:49.26 | *** join/#asterisk opus_ (i=opus@pabstblueribbon.net) |
21:49.28 | [TK]D-Fender | saftsack : it wasn, but many of their goals will be acheived under FreeSwitch, so it became a moot point I would imagine. So much more productive and fun building something nice than fixing up something broken. |
21:49.38 | mishehu | jm|work: ewww! no colonscopies in the channel please! |
21:49.42 | jm|work | ewww |
21:49.44 | jm|work | stop it! |
21:49.58 | jm|work | I meant instead of a semi-col .... no, that's as bad |
21:50.01 | *** join/#asterisk Scrye (n=ryan@2001:470:1f00:2514:280:c8ff:fec9:96d8) |
21:50.02 | xachen | hrm. I seem to sell lots of Openvox cards. Not many Digium though :( |
21:50.09 | Scrye | is there a url for supported phones? |
21:50.12 | Dr-Linux | opus_: your caps lock :) |
21:50.13 | VoIPMasta | jm|work: remove the "if you have one Wildcard..." text |
21:50.18 | jm|work | VoIPMasta: just doing that :) |
21:50.37 | Corydon-w | mishehu: so, you're still working hard on openpbx? When's the milestone that was due 6 months ago going to be released? |
21:51.07 | *** join/#asterisk h0 (n=h0@unaffiliated/fakhir) |
21:51.17 | jm|work | VoIPMasta: same error :( |
21:51.36 | mishehu | Corydon-w: you REALLY need to screw your head on a bit tighter. you haven't gotten a single fact right yet. I *never* was a developer for openpbx. Neither was anthm. And I *am* a developer for a module in freeswitch. |
21:51.51 | piwi | VoIPMasta: I installed linphone, changed its port but I cant login:'( SIP Identity is sip:username and SIP Proxy is sip:localhost isnt it? |
21:52.02 | VoIPMasta | yup |
21:52.12 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
21:52.12 | mishehu | and I'm doing it in C++ and maybe I'll dedicate it to kram, since he was rather insulting to me at last year's cluecon. |
21:52.14 | VoIPMasta | piwi: have you created a user in your sip.conf? |
21:52.23 | piwi | VoIPMasta: yes I did |
21:52.35 | VoIPMasta | piwi: what does asterisk say when you try to login? |
21:52.39 | Corydon-w | mishehu: contact me when freeswitch is finally ready in about 5 years. |
21:52.39 | VoIPMasta | or to "register" |
21:52.54 | mishehu | Corydon-w: that's about how long it took asterisk, no? |
21:52.56 | Scrye | has anyone used a UTStarCom F1000 with any success? |
21:53.00 | piwi | VoIPMasta: shall we go /msg? |
21:53.05 | Cresl1n | mishehu: I'm sure you NOTHING provocate it either |
21:53.20 | VoIPMasta | sure |
21:53.22 | mishehu | Cresl1n: I honestly did not understand what you said. |
21:53.22 | Corydon-w | mishehu: No, I learned about Asterisk after about 3. |
21:53.38 | mog | hey now everybody lets settle down a bit |
21:53.41 | mishehu | Corydon-w: learned about, but didn't do anything for about 5. |
21:53.44 | Cresl1n | heh |
21:54.02 | mog | gonna start kicking people, im looking at you Cresl1n ^_^ |
21:54.02 | Corydon-w | mishehu: No, I started contributing to Asterisk after 3 |
21:54.17 | hads | Yay. It's #basheveryone |
21:54.18 | mishehu | Corydon-w: so, does freeswitch step on your ego? because you're quite critical about a project that is around 6 months old. |
21:54.23 | MikeJ[Laptop] | ding ding ding.. corners everyone.. |
21:54.39 | jm|work | VoIPMasta: you want an account? |
21:54.41 | mog | hey everybody i have a problem with my x100p, discuss |
21:54.43 | jm|work | would that help? |
21:54.44 | Strom_C | my penis is bigger than your penis |
21:54.49 | MikeJ[Laptop] | mog, heh... |
21:54.55 | hads | lol mog |
21:55.00 | Cresl1n | Strom_C: oh no you didn't! |
21:55.02 | Corydon-w | No, I simply find it interesting that the freeswitch developers hang around the Asterisk channel trying to poach users |
21:55.11 | *** part/#asterisk h0 (n=h0@unaffiliated/fakhir) |
21:55.16 | mog | yes its all interesting, but back to my x100p |
21:55.16 | mitcheloc | mog, i have an x100p, what's the problem? |
21:55.27 | mishehu | mog: Hi, this is Eddie, your shipboard computer... |
21:55.28 | MikeJ[Laptop] | caller id problems? |
21:55.29 | mog | well when i plug my phone into it |
21:55.32 | VoIPMasta | hold on, I'm looking for one of my old zapata.conf files |
21:55.35 | mog | it doesnt give me dial tone |
21:55.36 | Cresl1n | it is quite funny how they can't seem to spend any of their time in their own chat rooms |
21:55.47 | jm|work | VoIPMasta: if you mean me: thanks. |
21:55.53 | Strom_C | i have an x100p that a dog pissed on and then a truck ran over and now it doesnt seem to work !!!! help plzkthx |
21:55.57 | Strom_C | mog mog mog |
21:56.00 | mishehu | Corydon-w: Just wait until we start launching missiles and declaring jihad on you. |
21:56.02 | mitcheloc | mog, i think, you are supposed to plug your phone service into the X100P, i'm not sure, turn to your right and ask kp, he might know ;) |
21:56.04 | angler | mog, lol |
21:56.10 | VoIPMasta | jm|work: yes, that was for you |
21:56.13 | mishehu | we will wipe out those asterisk infidels. ;-) |
21:56.19 | mishehu | </sarcasm> |
21:56.21 | mog | hmm im not sure |
21:56.28 | mog | i think it should work |
21:56.31 | Cresl1n | mishehu: oooh.... I'm sure all the secret NSA IRC spies just loved hearing that from you |
21:56.33 | mog | i mean it worked on my last x100p |
21:56.38 | MikeJ[Laptop] | mog.. I think that's a #asterisk-dev question no... |
21:56.53 | mog | heh |
21:56.53 | MikeJ[Laptop] | wait.. is it a digium x100p? |
21:56.57 | mog | well they sent me here |
21:57.06 | mog | its a clone ,,,,, err digium one |
21:57.09 | mitcheloc | Crestln: you didn't know that #asterisk is a sleepr cell? |
21:57.17 | hads | Ask jbot, he might know. Someone was trying to talk to him yesterday. |
21:57.22 | mishehu | Corydon-w: I am critical of asterisk because I use it and have experience with it. If you have such a problem with freeswitch users using asterisk as well, and mentioning the f word on the channel, maybe you should see if you can ban us all. |
21:57.24 | MikeJ[Laptop] | yeah.. brand new from digium last month right? |
21:57.27 | mitcheloc | Cresl1n, supposedly we are all going to take over the telecom industry... |
21:57.37 | Strom_C | i thought mine was an official digium clone of the clone of the official digium card |
21:57.42 | mitcheloc | * in the name of Allah |
21:57.47 | Dr-Linux | mishehu: how you know this word "Jihad" ? :S |
21:58.01 | MikeJ[Laptop] | mog :P |
21:58.08 | mog | okay seriously |
21:58.14 | Strom_C | Dr-Linux: I don't think there's a single person in north america that DOESN'T know what a jihad is :) |
21:58.17 | mog | everyone drop it |
21:58.33 | mog | there is another channel for everyone involved |
21:58.34 | mog | #drama |
21:58.35 | MikeJ[Laptop] | mog, I tried too.. no one listens to me either |
21:58.38 | mog | take it there |
21:58.43 | Corydon-w | mishehu: and I'm critical of freeswitch because it's vaporware. Free speech |
21:58.53 | MikeJ[Laptop] | mog.. join me? |
21:58.53 | Dr-Linux | Strom_C: aww but does it mean in english? i thought it's my language word? |
21:59.09 | jm|work | VoIPMasta: here's some dmesg shizzy if it helps: http://pastebin.ca/91763 |
21:59.13 | Strom_C | Dr-Linux: it's a loanword |
21:59.24 | mishehu | Corydon-w: hahaha and opinions are like assholes, and all stink except for mine. |
21:59.28 | mitcheloc | is a jihad a type of drink? |
21:59.35 | mog | anyone who wants to discuss this further go to #drama |
21:59.38 | mog | please |
21:59.49 | TommyTheKid | thanks for the sangoma reference guys! Solaris drivers, 8 port, 2U PCI cards.. I am in heaven :) |
21:59.50 | mishehu | Corydon-w: if anything, you only make us strive harder to make sure that freeswitch has a long, hardy life. |
21:59.53 | MikeJ[Laptop] | all the cool people are there! |
21:59.59 | MRH2 | this channel has been flagged for analysis following a keyword occurence ;) |
22:00.08 | MikeJ[Laptop] | heh |
22:00.08 | Corydon-w | mog: okay, I'm done now |
22:00.14 | mog | k |
22:00.16 | mog | next person |
22:00.17 | mitcheloc | but only MikeJ and mog are in #drama, meh |
22:00.19 | mog | is getting kicked |
22:00.21 | mishehu | mog: do I get the part? *grin* |
22:00.28 | mishehu | I've been working on my acting skills. |
22:00.34 | TommyTheKid | mmm lunch |
22:00.36 | *** part/#asterisk TommyTheKid (n=tommythe@mpk-edge.cto.sunit.net) |
22:00.43 | Dr-Linux | mitcheloc: Jihad is a kinda in the way of Allah |
22:00.48 | andrejkw | Is there any way of getting the amount of money in my account using Asterisk? |
22:01.03 | bkw_ | FreeSwitch IS NOT vaporware |
22:01.10 | Cresl1n | psssh |
22:01.14 | Cresl1n | just like openpbx |
22:01.15 | *** kick/#asterisk [bkw_!i=ejabberd@68.62.237.103] by mog (mog) |
22:01.18 | Cresl1n | and 16khz audio |
22:01.22 | *** kick/#asterisk [Cresl1n!i=ejabberd@68.62.237.103] by mog (mog) |
22:01.22 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
22:01.25 | mog | seriously |
22:01.26 | mog | stop it |
22:01.28 | mog | everyone |
22:01.33 | mog | im not kidding |
22:01.35 | mog | im tired of it |
22:01.36 | bkw_ | I just corrected the facts |
22:01.37 | MikeJ[Laptop] | yay.. |
22:01.39 | andrejkw | Can someone help, please? |
22:01.43 | MikeJ[Laptop] | sure |
22:01.46 | mishehu | mog: can I have one too? I feel left out. *grin* j/k |
22:01.51 | mog | sure |
22:01.52 | MikeJ[Laptop] | what ya need andrejkw |
22:01.58 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
22:01.58 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
22:01.58 | *** kick/#asterisk [mishehu!i=ejabberd@68.62.237.103] by mog (he asked for it) |
22:01.59 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) |
22:02.01 | [TK]D-Fender | ok, heading off to class, back in several hours |
22:02.04 | *** mode/#asterisk [+b %bkw_!*@*] by Corydon-w |
22:02.04 | znoG | mishehu: i can provoke you if you like ;) |
22:02.06 | mishehu | mog: I was just kidding. ;-) |
22:02.10 | mog | ^_^ |
22:02.13 | mog | just a joke |
22:02.18 | mog | but seriously no more drama here |
22:02.21 | andrejkw | Is there any way to find out the amount of money in my SIP account using Asterisk? |
22:02.22 | mog | we have a new channel for that |
22:02.24 | mog | #drama |
22:02.25 | *** part/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
22:02.27 | mog | feel free to join |
22:02.29 | mog | all welcome |
22:02.29 | Cresl1n | #asterisk-drama |
22:02.32 | Cresl1n | :-P |
22:02.32 | znoG | andrejkw: eh!? |
22:02.34 | *** mode/#asterisk [-b %bkw_!*@*] by Corydon-w |
22:02.38 | *** join/#asterisk AvoidingDeadlock (n=bkw_@adsl-70-143-63-127.dsl.tul2ok.sbcglobal.net) |
22:02.38 | mog | no just #drama |
22:02.40 | MikeJ[Laptop] | no.. #drama |
22:02.41 | andrejkw | I am making an IVR. |
22:02.41 | mog | is generic |
22:02.42 | Dr-Linux | mog: why you are kicking ??? |
22:02.53 | mog | too much drama |
22:02.55 | drray | you could do like the list, and say that question belongs on the drama list |
22:02.56 | mog | everyone was warned |
22:02.59 | xachen | op abuse imho :( |
22:03.00 | mog | several times |
22:03.11 | VoIPMasta | jm|work: can you hold on for a couple minutes, let me boot up my laptop, I'm sure I have some conf files there |
22:03.24 | jm|work | VoIPMasta: of course! Thank you very much :D |
22:03.27 | andrejkw | Is it possible to get the credit in the account? |
22:03.32 | andrejkw | Somehow? |
22:03.38 | VoIPMasta | andrejkw: what kind of account? |
22:03.46 | andrejkw | VoipBuster |
22:03.46 | jm|work | I should have said sulks _patiently_ :) |
22:03.51 | andrejkw | SIP |
22:04.17 | *** join/#asterisk foo (n=foo@unaffiliated/foo) |
22:04.19 | VoIPMasta | mmm you have to figure out a way to connect to voipbuster's database or to read data directly from their website |
22:04.19 | Dr-Linux | mog: you was also involved in asterisk/freeswitch discussion .. and when you didn't like you warned and kicking .. |
22:04.23 | foo | What is the status on 911 calls with asterisks? |
22:04.30 | foo | That is, knowing the location. |
22:04.52 | Corydon-w | foo: it's not something we can know |
22:05.09 | MikeJ[Laptop] | Cresl1n, where are you.... |
22:05.12 | MikeJ[Laptop] | come play! |
22:05.15 | MRH2 | anyone got issues with zaptel 1.2 since approx week commencing 3rd July? |
22:05.17 | file | it's a trick |
22:05.18 | file | don't do it |
22:05.23 | TripleFFFF | hmm good idea ill add this api to us |
22:05.29 | MikeJ[Laptop] | DO IT! |
22:05.33 | foo | Corydon-w: Hm, so there is no solution if someone calls 911 from a VoIP? That is, there is no location-specific notification 911 call centers get... |
22:05.44 | TripleFFFF | yeah |
22:05.51 | Corydon-w | foo: well, there is, but it's probably wrong |
22:05.56 | foo | ahh, I see. |
22:06.02 | TripleFFFF | costs a leg and an arm + an eye if you roll a 1 or a 2... ;) |
22:06.12 | Corydon-w | foo: you're need to set up something specific to handle that |
22:06.31 | foo | I see, but there is a way to handle that kind of situation? |
22:06.36 | MRH2 | foo: I route emergency calls over the pstn voip for everything else |
22:06.44 | TripleFFFF | MRH2 |
22:06.51 | TripleFFFF | then you will get the bills if one dial 911 to fuck around |
22:06.56 | Corydon-w | foo: there are various SIP providers who will complete the 911 call appropriately, but you must register the location of each phone in advance |
22:06.59 | TripleFFFF | 50 to 150 per call.. |
22:07.00 | MikeJ[Laptop] | Cresl1n, come back! |
22:07.09 | foo | Corydon-w: I see |
22:07.23 | [andromeda] | Is there a VoIP provider that will offer a free inbound PSTN phone number, that i will be able to use with asterisk? |
22:07.24 | mishehu | foo: where's your friend bar? |
22:07.39 | foo | mishehu: We were separated at birth. If you find him let me know. |
22:07.40 | MRH2 | <PROTECTED> |
22:07.58 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
22:07.58 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
22:07.59 | mishehu | foo: heh, good one ;-) |
22:08.22 | foo | :) |
22:08.30 | MRH2 | each company site has a line for it |
22:08.34 | foo | I see. |
22:08.41 | foo | Hm, it would be awesome if I could set up asterisk at home. |
22:08.42 | piwi | [andromeda]: I dont think such a provider exists... if it does please let me know!! |
22:09.26 | VoIPMasta | jm|work: /msg me |
22:09.36 | VoIPMasta | jm|work: so that we don't flood the chnnl |
22:09.39 | jm|work | :) |
22:11.36 | MRH2 | anyone using a 4xx card on a dell 2650 per chance? |
22:12.33 | NDT | 2 x 410s on a 2850 |
22:13.12 | MRH2 | cool ever get that flashing orange pci error on the box? |
22:13.15 | *** part/#asterisk Scrye (n=ryan@2001:470:1f00:2514:280:c8ff:fec9:96d8) |
22:13.48 | NDT | just when I had to disable one of the E1000 onboard nics to get the second card to work...cause they are an issue... |
22:14.18 | MRH2 | hmmm is that normally an irq type issue then? |
22:14.20 | anthm | I gotta go but before i do, corydon, say anything you like about freeswitch because it's your right, but do not pass me off as an ex-openpbx developer when you know for a fact that my name appears like 90 times in the asterisk code. You can not undo all the work i put into making asterisk get to where it is now. I am entitled to use it, work on it still, and to sit in this channel all i wish. |
22:14.21 | Dr-Linux | why asterisk doesn't have chan_sccp? :S |
22:14.43 | Strom_C | Dr-Linux: /me points at qwell |
22:14.54 | anthm | I still recall you deeply insulting kpflemming when pissed you off then the next day you are a cheerleader cos they give you commit access that is not very cool and i would have expeceted better. |
22:15.05 | NDT | MRH2: Normally... |
22:15.22 | MRH2 | reason being i get that orange pci error all the time with the recent zaptel 1.2 |
22:15.22 | Corydon-w | anthm: eh? |
22:15.36 | anthm | i've been reading this flame war |
22:15.45 | NDT | whats lspci say? |
22:15.52 | anthm | and i am not impresses with the ex-openpbx business |
22:15.53 | MRH2 | had to downgrade to svn from 2nd July to get rid of it |
22:16.17 | NDT | WHen I had the issue the 410 would show up as unknown communication device |
22:16.20 | anthm | any i am late, bbl |
22:16.25 | Corydon-w | I told mog I'm done, so there's nothing more to say on channel |
22:16.33 | *** join/#asterisk enjay- (n=enjay@71.216.165.97) |
22:16.51 | MRH2 | yep me too |
22:17.12 | Qwell[] | Well that was...interesting |
22:17.18 | Qwell[] | What DID I miss? |
22:17.25 | NDT | Going to have to play in the BIOS then with the IRQs heh |
22:17.33 | file | Qwell[]: lots |
22:17.34 | mishehu | Qwell[]: nothing worth the read. |
22:17.40 | file | the world exploded... |
22:17.46 | Qwell[] | file: ooo |
22:17.47 | file | I became the overlord of the planet... |
22:17.57 | Qwell[] | and then? |
22:18.05 | mishehu | file: and then I deposed you and reinstated the previous world order |
22:18.12 | MRH2 | yep i just couldn't seem to get rid of it with recent zaptel |
22:18.13 | opus_ | hey i'm an "ex" openpbx developer! :) |
22:18.14 | mishehu | just by overwriting you |
22:18.30 | MikeJ[Laptop] | opus_, there a lot of those? |
22:18.31 | Qwell[] | jbot: logs |
22:18.32 | jbot | apt/ibot/jbot/purl all log to http://ibot.rikers.org/<channelname>/ where channelname is html encoded ie: %23debian | lines that start with a space are not shown | some channels have stats at http://ibot.rikers.org/stats/<channelname>.html.gz, or updated "nightly" |
22:18.38 | opus_ | so osoma yo momma or whatever |
22:18.42 | Qwell[] | :p |
22:18.48 | MikeJ[Laptop] | hehe |
22:18.56 | drray | barrack yo momma oboma? |
22:19.40 | mishehu | drray: heh |
22:20.07 | Dr-Linux | why JerJer don't come to this channel? |
22:20.15 | MRH2 | I don't suppose that is bug report worthy is it? |
22:20.21 | Dr-Linux | even every asterisk user loves him? :) |
22:20.55 | MikeJ[Laptop] | #drama ! |
22:21.02 | NDT | MRH2: No...Dell says it's not their problem...Digium will say it isn't theirs, Sangoma will say you won't have the issue heh |
22:21.10 | MRH2 | lol |
22:21.56 | mog | whats the problem MRH2 ? |
22:22.09 | *** join/#asterisk AJaymn (i=AJmn@70.59.126.197) |
22:22.40 | MRH2 | looks like wierd irq conflict with a 4xx card and zaptel1.2 > 3rd July |
22:22.55 | Corydon-w | MRH2: is the board still taking interrupts? |
22:23.34 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
22:23.34 | *** mode/#asterisk [+o mog_home] by ChanServ |
22:23.54 | jm|work | I need to back up some config files |
22:24.21 | MRH2 | well it seemed to work just came up with a "dell orange flashing pci error" on the chasis |
22:24.26 | *** join/#asterisk lullabud (n=lullabud@12.24.42.67) |
22:24.38 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
22:24.54 | MRH2 | since moved back to the 2nd July version and that works as it always has done |
22:25.02 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
22:25.14 | Corydon-w | On the motherboard or on the Digium card? |
22:25.14 | lullabud | does anybody know of a download link for the Polycom 3.1.2 bootrom? |
22:25.25 | MRH2 | on the motherboard |
22:25.39 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
22:25.39 | *** mode/#asterisk [+o Qwell[]] by ChanServ |
22:26.07 | Corydon-w | Does Dell have a diagnostic code for what that means? |
22:26.44 | type0 | you talking about a poweredge? |
22:26.55 | type0 | we had some blinking orange lights with our 4 poweredges |
22:27.19 | MRH2 | i'll check with dell on that - money seems to be on an irq issue |
22:27.25 | NDT | It's the E1000s and the Digium cards...they don't like to share interrupts |
22:28.24 | Corydon-w | MRH2: typically when we have an IRQ issue, we try either moving the card to a different slot, or failing that, we set noapic in the Linux kernel boot options |
22:28.27 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
22:28.43 | jm|work | oh; it not existing might be it |
22:28.50 | NDT | lol |
22:29.26 | Strom_C | haha |
22:29.34 | VoIPMasta | jm|work: however don't get too excited with chan_h323 since it isn't THAT good |
22:29.42 | MRH2 | yep tried the ususal stuff and couldn't get rid of it - just weird it happens only after a recent change to zaptel |
22:29.47 | jm|work | is it a m$ thing? |
22:29.48 | VoIPMasta | if JerJer reads this he'll start flaming me ;) |
22:30.10 | VoIPMasta | jm|work: no, $ms has nothing to do with h323 |
22:30.12 | lullabud | nevermind on the polycom boot rom. amazing what a google search including "index of" will get you. ;-) |
22:30.32 | jm|work | VoIPMasta: no I just read that: amazingly they conformed to a RFC for once :O |
22:30.48 | jm|work | well; an h. at least |
22:32.38 | Corydon-w | MRH2: dell.com suggests that you're experiencing a low power condition. Have you checked to make sure that you're getting sufficient voltage from both power supplies? |
22:32.39 | MRH2 | I'll try disabling everything i can on the board and work up from there. |
22:33.16 | NDT | MRH2: Heh ain't many things to disable in that bios heh |
22:33.34 | MRH2 | lol |
22:33.36 | NDT | Swicth the IRQ on the card in the bios to one the NICs aren't using |
22:33.47 | MRH2 | yep did that |
22:33.51 | jm|work | so apart from pulver who else should I subscribe to? |
22:34.02 | NDT | Worked for me...until I added another card heh |
22:34.06 | MRH2 | even tried moving the card |
22:35.03 | Corydon-w | MRH2: are there any events in the BIOS event log? |
22:35.04 | drray | move it to another PC? |
22:35.06 | *** join/#asterisk teknoprep (n=chris@unaffiliated/teknoprep) |
22:35.09 | teknoprep | hi all |
22:35.21 | NDT | Bottom slot worked perfectly for me...until I added that second card and had to disable one of the onboard NICS...needless to say...Anything I do with these poweredges we have again...will be with a sangoma card |
22:35.45 | jm|work | bah |
22:35.56 | jm|work | this was not the best day to leave my mobile phone at work by mistake |
22:36.02 | MRH2 | croydon i'll have to retest it with a later zaptel again - anything of note round about the 7th July that would mess with it? |
22:36.14 | VoIPMasta | jm|work: do you want to test your "incoming" calls? |
22:36.18 | jm|work | for some reason my land line is engaged every time I try to phone it .... |
22:36.23 | jm|work | VoIPMasta: perhaps :) |
22:36.31 | NDT | You will see references to those NICS and 405/410/411 cards in a few places... |
22:36.37 | jm|work | it's 11:40pm here and the GF is bed and the extensions will probably ring |
22:36.41 | VoIPMasta | jm|work: pm me your number and I can dial it (so that you can see if the call hits your asterisk box) |
22:36.49 | VoIPMasta | I see |
22:36.53 | jm|work | GFs, eh. |
22:36.57 | jm|work | Who'd 'ave 'em. |
22:37.13 | jm|work | "Still doing that boring phone stuff? I'm going to bed." |
22:37.18 | jm|work | (: |
22:37.47 | VoIPMasta | Start worrying when she decides to go to bed with someone else, since you are more into voip than into her |
22:37.50 | VoIPMasta | hehehe |
22:37.54 | jm|work | hehehe |
22:38.30 | jm|work | I'm pleased that I can dial out now :) |
22:38.48 | MRH2 | ok gtg - thanx all - i'll report back ;) |
22:38.52 | jm|work | that means that when I'm in New York on business (HAH!) I can call UK at national rates :) |
22:38.57 | NDT | later MRH2 |
22:39.23 | jm|work | I need more POPs! |
22:39.29 | jm|work | or PsOP |
22:39.31 | jm|work | or whatever |
22:40.17 | Dr-Linux | i'm sorry friend, i passed you wrong comments about kicking. |
22:41.36 | Dr-Linux | Qwell[]: you come late today? :) |
22:41.42 | Qwell[] | Dr-Linux: yeah :p |
22:41.53 | Qwell[] | had to get around the firewall again... |
22:42.44 | Dr-Linux | Qwell[]: i like some opss here, they never talk, but when they talk they give some good appriated help |
22:43.12 | Dr-Linux | Qwell[]: but today i saw something different |
22:43.24 | jm|work | 'mailbox' ... sheesh |
22:43.30 | Qwell[] | Dr-Linux: it's best to leave it alone... |
22:43.50 | Strom_C | jm|work: i've never heard it called "answerphone" in the U.S. |
22:43.59 | jm|work | "voicemail" |
22:44.08 | Strom_C | exactly :) |
22:44.12 | jm|work | forgive me for spawning you a language 1000 years ago ;) |
22:44.33 | *** join/#asterisk bkw_ (n=bkw_@asterisk/friend-and-developer/bkw) |
22:44.42 | Dr-Linux | Qwell[]: actually everyone have respect in his/her own hand, some people dont understand .. |
22:44.52 | Strom_C | Englsh in 1006 didn't exactly resemble modern English :) |
22:45.01 | Dr-Linux | Qwell[]: so when you are going to dedicate chan_sccp for asterisk? :P |
22:45.11 | jm|work | Nay, Siree. I bequeath thee, such! |
22:45.17 | Qwell[] | Dr-Linux: Never. I didn't write it |
22:45.51 | Dr-Linux | Strom_C: i don't know english, but today i say .. i'm better :P |
22:46.02 | jm|work | I wish the analogue phone converters were as cheap as my dog-pissed-on-van-run-over-card :( |
22:46.15 | Dr-Linux | Qwell[]: why asterisk doesnt'have chan_sccp? |
22:46.35 | Qwell[] | Strom_C: ? |
22:46.37 | Dr-Linux | Qwell[]: Strom_C pointed me to you |
22:46.42 | Corydon-w | Because skinny is the same thing |
22:47.04 | Corydon-w | and chan_skinny predates chan_sccp |
22:47.06 | Dr-Linux | Corydon-w: Qwell[] don't think |
22:47.07 | Qwell[] | trying not to say unkind words about (the MIA, possibly dead) Sergio... |
22:47.11 | Dr-Linux | doesn't think the same |
22:47.41 | Dr-Linux | Qwell[]: i think Sergio has died :S |
22:47.51 | Qwell[] | I'm sure he has :p |
22:48.04 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
22:48.12 | Dr-Linux | Qwell[]: also i guess he had some problem with Sourceforg ... |
22:48.15 | Qwell[] | would be pretty hard to get a disclaimer from a dead man... |
22:48.23 | Corydon-w | in which case the copyrights don't revert to the public domain for another 75 years |
22:48.39 | Qwell[] | Corydon-w: longer, if Disney has anything to say about it |
22:48.56 | Corydon-w | corporations get 95 years, period |
22:49.02 | Dr-Linux | Qwell[]: sorry i don't understand difficult english :S |
22:49.04 | Qwell[] | fun |
22:49.06 | Corydon-w | individuals get life plus 75 |
22:49.11 | Dr-Linux | the copyrights one |
22:49.13 | Qwell[] | life PLUS 75?! |
22:49.23 | Corydon-w | Yep |
22:49.26 | Qwell[] | pfft |
22:50.01 | Qwell[] | so, like... |
22:50.09 | Qwell[] | maybe somebody should find out if he is still actually alive |
22:50.25 | *** join/#asterisk Skarmeth (n=Skarmeth@201009018188.user.veloxzone.com.br) |
22:50.35 | Corydon-w | You could also get his heirs to disclaim the copyright |
22:50.38 | Dr-Linux | Qwell[]: if redhat wants, can they take my domain from me? |
22:50.45 | Qwell[] | Dr-Linux: this is true |
22:50.48 | TripleFFFF | hey |
22:50.48 | Qwell[] | erm |
22:50.52 | Qwell[] | Corydon-w: this is true |
22:50.57 | Qwell[] | Dr-Linux: What domain? Probably not |
22:51.05 | Dr-Linux | Qwell[]: redhat.pk |
22:51.09 | Qwell[] | heh |
22:51.16 | Dr-Linux | tele.pk |
22:51.20 | Qwell[] | possible |
22:51.22 | Dr-Linux | i don't think tele.pk |
22:51.25 | Corydon-w | Dr-Linux: it depends. How much money do you have to spend on a lawyer? |
22:51.26 | jm|work | VoIPMasta: is there an GUI softphone that doesn't have a bazzilion R-DEPS like gnome and shizzy? |
22:51.29 | Dr-Linux | but i'm asking about redhat.pk? |
22:52.02 | Qwell[] | Dr-Linux: That is likely a trademark issue (assuming they have a trademark there) |
22:52.03 | Dr-Linux | Corydon-w: how can i take stand with red hat? huh i'm a poor pakistani |
22:52.50 | Dr-Linux | Qwell[]: i'm asking bcoz i just got this domain, if they will take it from me, then i'm not gonna work on it |
22:52.51 | Corydon-w | They could try, but unless you're doing something that violates their trademark, they won't succeed unless they can successfully make you spend enough money that you just give up |
22:52.56 | Dr-Linux | then i'll prefer tele.pk |
22:53.06 | *** part/#asterisk hohum (n=dcorbe@12.195.58.235) |
22:53.19 | Corydon-w | And that's for any corporation, not just RedHat |
22:54.20 | Dr-Linux | Corydon-w: so will they give me money that i'd have spent on site, or i'll be just removed? |
22:55.57 | *** join/#asterisk dlynes_laptop (n=dlynes@S0106001217014b92.vc.shawcable.net) |
22:56.19 | Dr-Linux | bad bad :( i have about 15 trade mark domains |
22:56.46 | [andromeda] | Has anyone here used Gizmo's inbound service with asterisk? |
22:57.18 | Corydon-w | Dr-Linux: generally not, but talk to a lawyer to be sure |
22:58.35 | Dr-Linux | Corydon-w: we don't have tht system here, remember i'm a tribal guy .. no police, no courts, nor govt. etc rulse |
22:58.41 | Dr-Linux | rules |
22:59.54 | opus_ | yeah you should be fine in pakistan |
23:00.03 | Corydon-w | Does the tribal chief like you? |
23:00.12 | Qwell[] | more importantly... |
23:00.16 | Qwell[] | Does the tribal chef like you? |
23:00.24 | opus_ | why are so many VOIP people from pakistan :) |
23:00.30 | Dr-Linux | Corydon-w: sorry i didn't understand friend? |
23:00.36 | dlynes_laptop | opus_, call centers |
23:00.38 | drray | cheap |
23:00.39 | drray | er |
23:00.40 | dlynes_laptop | opus_, same for india |
23:01.02 | Dr-Linux | opus_: i think i'm only guy in tribals who knwos a bit Linux/voip etc |
23:01.08 | Corydon-w | Dr-Linux: if the system of justice is in your favor, then you have little to worry about |
23:01.43 | Dr-Linux | Corydon-w: but domains case is international |
23:02.10 | Dr-Linux | if they take my domain, i'll do again bad bad stuff as before |
23:02.11 | opus_ | red hat might not be registered to do business in your country :) so it might be fair game |
23:02.26 | Corydon-w | Dr-Linux: they still have to contact you. If the lawyer is strung up in the town square before he serves you with papers, there's not much they can do |
23:02.29 | opus_ | just like going to mcdonalds in Iran I think? :) Or Star burger in mexico |
23:02.45 | Dr-Linux | tribal people only know guns etc stuff |
23:02.45 | *** join/#asterisk gaupe (i=rmo@slogen.sunnmore.net) |
23:02.59 | opus_ | Dr-Linux is bin laden really dead? :) |
23:03.09 | dlynes_laptop | opus_, didn't you know? |
23:03.15 | dlynes_laptop | opus_, bin laden is Dr-Linux's cousin |
23:03.24 | Dr-Linux | opus_: if they can have after somethime .. |
23:03.38 | Dr-Linux | opus_: who says ben laden dead?? |
23:03.44 | opus_ | the president of pakistan said he was dead |
23:03.50 | Dr-Linux | i'm from the same town |
23:03.52 | NDT | Has anyone looked at the UNIQUEID of a call after it has returned from a macro? |
23:03.56 | dlynes_laptop | See???? |
23:03.58 | drray | I believe he's been dead for a year or two |
23:03.59 | Dr-Linux | opus_: lolzz |
23:04.03 | dlynes_laptop | He's Bin Laden's cousin!!!!! |
23:04.06 | drray | but I could be wrong |
23:04.08 | Dr-Linux | he is just a shit liar |
23:04.15 | opus_ | bin laden runs asterisk, i knew it. |
23:04.31 | opus_ | Dr-Linux the president? |
23:04.44 | Dr-Linux | he says, bcoz he don't want US army to get in tribals area of pakistan to search him |
23:04.47 | dlynes_laptop | Dr-Linux, he just wants the americans out of pakistan and afghanistan, so he can get on with his life :) |
23:04.49 | opus_ | Dr-Linux how tall is bin laden? |
23:05.03 | Dr-Linux | opus_: and US army make him an issue to try to get in tribals |
23:05.23 | opus_ | huh? |
23:05.39 | *** part/#asterisk foo (n=foo@unaffiliated/foo) |
23:05.40 | Dr-Linux | that hows they bombing my area on my Eid day :( my baby cousins were died :( |
23:05.52 | Qwell[] | Eid day? |
23:06.02 | dlynes_laptop | muslim holiday near christmas time |
23:06.07 | Dr-Linux | dlynes_laptop: there is no American army in tribal dude. |
23:06.10 | Dr-Linux | and can't be |
23:06.11 | dlynes_laptop | a time of celebrating by eating lots after a time of fasting |
23:06.24 | dlynes_laptop | eids |
23:06.25 | Dr-Linux | Tribals are fighting with Pakistan army |
23:06.54 | dlynes_laptop | right, Dr-Linux ? |
23:07.11 | Dr-Linux | opus_: actually Laden was not that much as US mentiond |
23:07.21 | Dr-Linux | dlynes_laptop: Eid is our holiday |
23:07.29 | dlynes_laptop | that's what i said |
23:07.35 | dlynes_laptop | It's like the muslim version of Christmas |
23:07.45 | dlynes_laptop | A day to pig out on lots of food |
23:07.47 | Dr-Linux | dlynes_laptop: imgine .. how US planes without Poilot bombing my area .. |
23:07.49 | dlynes_laptop | and exchange presents |
23:07.52 | Dr-Linux | yes yes |
23:08.00 | Dr-Linux | pig? aghhh |
23:08.44 | Corydon-w | Dr-Linux: it's a euphemism, meaning "to eat like Americans" |
23:09.02 | Dr-Linux | we don't eat pork |
23:09.11 | dlynes_laptop | nobody said anything about eating pork |
23:09.14 | Corydon-w | Dr-Linux: as in, a large quantity |
23:09.47 | Dr-Linux | sorry, dlynes_laptop some time i don't understand english but always try though |
23:09.53 | Qwell[] | "pig out" == "gorge" |
23:09.55 | Corydon-w | dlynes_laptop: you're talking to someone without much knowledge of euphemisms |
23:10.24 | dlynes_laptop | Dr-Linux, nod...thought you were jsut reading it wrong |
23:10.49 | Corydon-w | Hence, the English => Russian => English translation into "The vodka is good, but the meat is rotten." for "The spirit is willing, but the flesh is weak." |
23:11.04 | NDT | Qwell: You have any idea how I can work around the uniqueid of a call changing when returning from a macro? I need the original uniqueid for use again...but when you return from the macro it changes it |
23:11.36 | Dr-Linux | Corydon-w: it's not about Knowledge, it's bout culture and langauge. |
23:11.48 | *** join/#asterisk heliosj (n=jeff@pdpc/supporter/active/xheliox) |
23:11.56 | Dr-Linux | i'm sure you'd not know even a word of my langauge except Jihad |
23:11.59 | dlynes_laptop | Dr-Linux, yes, americans love to eat :) |
23:12.13 | *** part/#asterisk ariel_ (n=ariel_@74.8.35.2) |
23:12.32 | dlynes_laptop | Dr-Linux, jihad's an arabic word, not urdu, isn't it? |
23:12.37 | Dr-Linux | dlynes_laptop: that's how i can't go to USA/UK :) |
23:13.18 | Dr-Linux | dlynes_laptop: it's the Same, it's a Holly word |
23:13.53 | hads | dlynes_laptop: Got an encouraging email back from Sangoma today about organising telepermits for NZ. At this stage they are interested in getting it done. |
23:14.08 | dlynes_laptop | cool |
23:14.37 | Dr-Linux | dlynes_laptop: what you say, if i do somethign wrong like cybercrime, what FBI can do with me? |
23:14.53 | Corydon-w | Dr-Linux: you don't really want to know |
23:15.01 | Qwell[] | federal PMITA prison |
23:15.12 | dlynes_laptop | they'll sick the cia on you and supply you with kalishnikovs |
23:15.14 | Corydon-w | In any case, it's not the FBI that would do it to you |
23:15.37 | Dr-Linux | who will do then? |
23:15.47 | Corydon-w | Dr-Linux: even in the US, people disappear without a trace |
23:16.09 | *** join/#asterisk anthm (n=anthm@000-439-099.area4.spcsdns.net) |
23:16.09 | *** mode/#asterisk [+o anthm] by ChanServ |
23:16.21 | Dr-Linux | Corydon-w: correct |
23:16.39 | Corydon-w | Dr-Linux: I suspect the same is true of most countries |
23:17.02 | Dr-Linux | Corydon-w: but i dont think they can do anything in tribals |
23:17.33 | Corydon-w | Dr-Linux: so nobody you've ever known has ever disappeared? |
23:17.55 | kcortez | ] |
23:18.03 | Dr-Linux | Corydon-w: here? no |
23:18.14 | AndyCap | hads: they'll get to you when they're done with Australia. :P |
23:18.29 | hads | AndyCap: Heh :) |
23:18.35 | Dr-Linux | Corydon-w: i know a number of peoples doing carding/hacking etc and nothing happend |
23:18.35 | Corydon-w | Dr-Linux: I'm surprised, actually. I've known a few people who have vanished |
23:19.17 | gaupe | it happened a lot here in Norway too, but that was mostly between 1940 and 1945 ;) |
23:19.25 | Corydon-w | Dr-Linux: they may have disappeared of their own volition, or they may have been disappeared by someone else, and we'll probably never know the truth |
23:19.48 | Dr-Linux | Corydon-w: well, in the tribal there is some different case |
23:19.57 | Dr-Linux | Corydon-w: do you know about tribals? |
23:20.05 | Corydon-w | gaupe: these are all in the last 10 years, for people who, as far as I'm aware, have never left the USA |
23:20.26 | Dr-Linux | Corydon-w: even Pakistan police can't enter in the tribals area |
23:20.27 | *** join/#asterisk kfudge (n=kzymxa@24-217-137-21.dhcp.stls.mo.charter.com) |
23:20.30 | AndyCap | Corydon-w: at least not before they disappeared.. ;) |
23:20.47 | kfudge | how would I make an extension say 99 go to an ivr |
23:20.54 | AndyCap | extraordinary rendition and all that |
23:21.00 | Corydon-w | AndyCap: well, they could be anywhere. That's part of the point. |
23:21.17 | gaupe | Dr-Linux: we've got pakistani tribes in oslo too |
23:21.45 | Corydon-w | AndyCap: the two people I'm thinking of in particular are both white adults, one Christian, one Jewish. |
23:22.07 | Dr-Linux | gaupe: oslo? |
23:22.13 | dlynes_laptop | Dr-Linux, norway |
23:22.26 | Dr-Linux | awww |
23:22.54 | Dr-Linux | everything is broken here related to norway/danmark |
23:23.06 | gaupe | Dr-Linux: gangs made of pakistanis battling for the control of the illegal business |
23:23.09 | Strom_C | good things from |
23:23.25 | Strom_C | er |
23:23.30 | Strom_C | good things from Norway: a-ha :) |
23:23.35 | Dr-Linux | gaupe: we are really very unlucky, you know why? |
23:23.47 | gaupe | Dr-Linux: no? |
23:23.59 | Dr-Linux | gaupe: my few of family in Pakistan army, |
23:24.11 | Dr-Linux | gaupe: and my rest of family is tribals |
23:24.47 | Dr-Linux | and currently pakistan army and tribals are fighting bad, killing each other, tribals have power.. |
23:25.00 | *** join/#asterisk lyroy (n=lyroy@modemcable146.87-83-70.mc.videotron.ca) |
23:25.20 | Dr-Linux | then imagine .. an army guy is attacking at his own home |
23:25.32 | lyroy | Does someone here ever configure a Cisco ATA 186 v.3.1.. SIP with Asterisk? |
23:25.32 | Dr-Linux | gaupe: why? do you know |
23:27.38 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
23:32.02 | *** join/#asterisk jbot_ (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
23:32.02 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.10 and Zaptel 1.2.7 released! (July 14, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx |
23:32.14 | Dr-Linux | dlynes_laptop: you are good in work, or good in making deals with customers? :) |
23:32.27 | *** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com) |
23:32.31 | dlynes_laptop | I'm good at the work; the owner is good at finding customers |
23:33.01 | Dr-Linux | good match |
23:33.19 | Dr-Linux | dlynes_laptop: how many peoples are working at your work place? :) |
23:33.21 | dlynes_laptop | well, the better match is that he's a traditional telephone guy, and i'm a computer guy |
23:33.27 | dlynes_laptop | so it covers all the bases |
23:33.39 | dlynes_laptop | right now, two of us for the most part |
23:33.42 | riddlebox | what could cause my asterisk to not give me ringback some of the time(mostly to pstn) cell phones work fine? |
23:33.48 | dlynes_laptop | however, we do subcontract once in a while |
23:34.12 | dlynes_laptop | riddlebox, mind rephrasing that? it didn't make any sense |
23:34.14 | Dr-Linux | dlynes_laptop: what you guys offer? |
23:34.25 | dlynes_laptop | Dr-Linux, everything but the kitchen sink |
23:34.32 | Dr-Linux | :S |
23:35.27 | riddlebox | dlynes_laptop, when I call someone lets say a home phone, I have no ringback, but if I call a cellphone I get ringback |
23:35.53 | dlynes_laptop | by ring back, you mean a ringing sound while you're waiting for the other end to answer? |
23:36.00 | *** part/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
23:36.36 | dlynes_laptop | riddlebox, ? |
23:36.40 | riddlebox | dylnes_laptop,yes |
23:36.52 | dlynes_laptop | riddlebox, is the call to the cellphone going through a gsm gateway? |
23:37.01 | riddlebox | dylnes_laptop,yes |
23:37.14 | dlynes_laptop | riddlebox, and the call to the phone is going through BRI ISDN? |
23:37.18 | Dr-Linux | dlynes_laptop: he means, >> dialing number >> dialed >> now listening and waiting >>> toooooon tooooooon .... tooooooooon ... toooooon |
23:37.19 | Dr-Linux | :) |
23:37.23 | lyroy | Does someone here is using a Cisco ATA 186 if so what version are u using (SIP)? |
23:37.38 | wunderkin | tooon? |
23:37.44 | wunderkin | sounds different there |
23:37.56 | riddlebox | dlynes_laptop, it is through broadvoice |
23:37.57 | dlynes_laptop | wunderkin, the ringing in india sounds super weird, too |
23:38.13 | Dr-Linux | wunderkin: what's there sound? |
23:38.16 | dlynes_laptop | riddlebox, maybe they're not passing progression |
23:38.22 | dlynes_laptop | riddlebox, try complaining to them |
23:38.36 | dlynes_laptop | riddlebox, i.e. call progression |
23:44.24 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
23:44.24 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org -=- Asterisk 1.2.10 and Zaptel 1.2.7 released! (July 14, 2006) -=- FreePBX/AMP/Asterisk@Home/Trixbox support in #freepbx |
23:48.29 | *** join/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
23:54.53 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
23:55.32 | *** join/#asterisk carl0s- (n=carl0s@compsup.demon.co.uk) |
23:55.47 | carl0s- | msg nickserv identify abracadabra |
23:56.10 | dlynes_laptop | carl0s-, thanks for telling us what your nickserv password is |
23:56.11 | carl0s- | :p |
23:56.18 | Dr-Linux | carl0s-: lolz |
23:56.29 | carl0s- | :D |
23:56.36 | Dr-Linux | carl0s-: do like this: |
23:56.38 | carl0s- | I've been wanting to do that for a while :D |
23:56.47 | Dr-Linux | /ns identify abracadabra |
23:57.21 | dlynes_laptop | Dr-Linux, /ns boot dr-linux |
23:57.29 | carl0s- | Dr-Linux: what does that do? |
23:57.52 | carl0s- | It wasn't really my nickserv password. I just pretended for a laugh. |
23:57.58 | Dr-Linux | carl0s-: same that you were trying to doing, but in short way |
23:58.01 | carl0s- | (yeah like people are going to beleive me) |
23:58.07 | carl0s- | Dr-Linux: ah, thanks :) |
23:58.21 | carl0s- | I think I probably wouldn't have been able to join the channel if I hadn't already identified anyway |
23:58.32 | Dr-Linux | carl0s-: like what it does: |
23:58.39 | Dr-Linux | //say $OS |
23:58.57 | carl0s- | yep, I've seen that stuff before. a long time ago. |
23:59.08 | dlynes_laptop | $OS |
23:59.10 | dlynes_laptop | and? |
23:59.30 | Dr-Linux | dlynes_laptop: you are not on windows :) |
23:59.38 | dlynes_laptop | yes, i am |
23:59.42 | dlynes_laptop | windows xp |
23:59.49 | carl0s- | hmm I thought it'd report the O/S. I remember people trying to get newbies to do things like "/exec $decrypt[a83hs83hfd893h93]" where it decrypted to "rm -rf /" |
23:59.54 | carl0s- | XP |