00:00.03 | sivana | brc_: cannot create_adm_p /tmp/cvs-serv32470/include/solaris-compat/sys |
00:00.09 | sivana | not sure if that's helpful |
00:00.09 | brc_ | yup |
00:00.21 | Chuji | Ok, 6:00, just started my make on Asterisk with a P133. Let's see how long this takes |
00:00.34 | Chuji | Damin: Don't you use a 133 too? |
00:02.33 | sivana | brc_: did you fix? |
00:02.38 | brc_ | sivana, I can't fix |
00:02.44 | sivana | brc_: its working now |
00:02.46 | brc_ | just going to figure out who's it is and tell them |
00:02.54 | brc_ | I think the same one keeps breaking |
00:02.54 | sivana | unless I hit a mirror |
00:03.02 | brc_ | use the ip |
00:03.06 | brc_ | instead of cvs. |
00:03.08 | brc_ | to make sure |
00:03.20 | sivana | probably bkws.. hehe |
00:03.24 | *** join/#asterisk eXoR` (~exor@xdsl-213-196-200-5.netcologne.de) |
00:03.26 | brc_ | yes |
00:04.23 | *** join/#asterisk Sedorox (~Sed@Neptune-W.client.wlgrv.pa.sed6.net) |
00:05.21 | jesster | anyone know if SetCallerPres works on sip ? i don't see it working for me. Im doing a call to another Sip provider |
00:05.34 | Chuji | sheesh, g729 licensing is a bitch |
00:05.38 | brc_ | yup |
00:05.52 | Chuji | Every time I monitor, or send to vm, or anything |
00:05.57 | Chuji | I need another damn license |
00:06.01 | Chuji | That's crazy |
00:06.06 | brc_ | no no no no |
00:06.10 | brc_ | you need a license per channel |
00:06.33 | Chuji | if I'm monitoring a g729 call saving to wav, I need two licenses |
00:06.40 | brc_ | yup |
00:06.51 | bparker | does that translate to a license per conconcurrent call |
00:06.52 | brc_ | I don't really see a better way to do it then per channel |
00:06.55 | brc_ | bparker, no |
00:06.59 | Chuji | Or if a g729 call goes to voicemail, I need two |
00:07.03 | *** part/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
00:07.09 | brc_ | Chuji, you sure about that? |
00:07.26 | brc_ | if you've got a g729 phone, that calls a number then goes to leave a message I believe that only takes one |
00:07.28 | brc_ | one channel... |
00:07.43 | Chuji | From Digium support: |
00:07.44 | Chuji | Voicemail going to G.729 will use a license. |
00:07.44 | Chuji | The sames goes for flashing over on a call using G.729. This will us an |
00:07.44 | Chuji | additional license. |
00:07.44 | Chuji | If you are using the monitor application to record the call to something |
00:07.44 | Chuji | other than G.729 this will use a license as well. |
00:08.06 | brc_ | isn't that what I just said? |
00:08.21 | brc_ | one license per channel up to vm from a g729 device |
00:08.38 | Chuji | So if they leave a message, only one license? |
00:09.01 | bparker | a channel = a device? |
00:09.05 | *** join/#asterisk lesouvage (~erik@cc341200-a.assen1.dr.home.nl) |
00:09.15 | brc_ | no, not at all |
00:09.16 | *** join/#asterisk cripito (~saul@c-65-34-156-173.hsd1.fl.comcast.net) |
00:09.16 | Chuji | yeah sip/ |
00:09.23 | cripito | hi |
00:09.45 | brc_ | bparker, a channel can be thought of as a pipe to somewhere |
00:09.53 | brc_ | that could be another device, or voicemail, or whatever |
00:09.54 | bparker | please explain what a channel is. I am about to have to license g729 so I need to know this |
00:10.08 | brc_ | I just did |
00:10.08 | bparker | oh ok |
00:10.17 | bparker | yeah I just saw that |
00:10.48 | Chuji | Guess I'm not going to freely use g729 like I thought |
00:10.57 | Chuji | especially since I was going to monitor each call |
00:11.08 | Chuji | Have to buy 2 per user |
00:11.17 | brc_ | if you are monitoring a call between two g729 devices, the monitor app gets stuck in the middle, with a pipe from g729 device a to monitor, then from monitor to g729 device b |
00:11.20 | Nugget | I have never encountered anything in asterisk usage that has made me wish I had g729 available to me. What am I missing? |
00:11.22 | bparker | so when voicemail is concerned, voicemail = 1 channel no matter how many calls are going into it at a given time |
00:11.29 | brc_ | Nugget, certian devices |
00:11.31 | drumkilla | just don't use 729 :) |
00:12.01 | Chuji | Nugget : I've got some dsl clients that have terrible upstream |
00:12.02 | Chuji | lots of jitter |
00:12.08 | Nugget | *nod* |
00:12.10 | brc_ | is that what I just said? no, it is not. I said a channel is a pipe from somewhere to somewhere, so if you have two devices called in to voicemail you will obviously need two pipes |
00:12.14 | Chuji | gsm is OK, but 729 clears it up |
00:12.26 | drumkilla | I guess I'm just cheap |
00:12.28 | drumkilla | so I just use gsm |
00:12.40 | bparker | ok I think I follow now |
00:12.41 | brc_ | MOST devices do not support gsm |
00:12.42 | Chuji | I may have to go to gsm on some of them |
00:13.08 | Chuji | brc_ : Good call, I'll have to look at that |
00:13.15 | brc_ | sides, speex > gsm :) |
00:13.20 | drumkilla | most of my voip stuff is just trunking to other boxes |
00:13.23 | lesouvage | I have installed sipp (a sip stresstest tool) on a linux box. It runs but is not connecting to my asterisk box. What do I have to do to really stress my *box? |
00:13.24 | Chuji | brc_ : I've been using sipura 2000 for this testing |
00:13.38 | drumkilla | yeah, it's cool how you can tweak speex |
00:13.41 | brc_ | yup |
00:14.10 | file[laptop] | how are you two doing? |
00:14.20 | brc_ | I'm doing TERRIBLE |
00:14.23 | drumkilla | working as usual |
00:14.25 | Chuji | brc_ : So you would use speex over gsm? |
00:14.27 | drumkilla | I'm doing GRRREAT |
00:14.51 | brc_ | hard to say...probably if it was available since you can tune it to your needs |
00:15.03 | brc_ | (asterisk to asterisk...) |
00:16.29 | jayeola | going for my 1st compile of asterisk. anyone tested this with sflphone? |
00:16.31 | Nugget | http://lnk.nu/slacker.com/na.cgi <-- heh, you can see my conference call this afternoon |
00:17.05 | *** join/#asterisk dersteer (~travis@24.231.151.119.gha.mi.chartermi.net) |
00:17.23 | lesouvage | screen 4 of sipp says "no action found on any messages". |
00:19.42 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
00:20.28 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
00:24.39 | FuriousGeorge | hey guys. i recently managed to get asterisk somewhat configured and more or less starting up with no problems. now im trying what i think is the next logical step: connecting a client to make outbound calls |
00:24.48 | FuriousGeorge | for some reason i cant authenticate |
00:25.02 | FuriousGeorge | and i know its not b/c i have the username or pw i set in sip.conf wrong |
00:25.28 | Nugget | what does the asterisk console say when it fails? |
00:26.00 | elriah | Anyone here having problems with agi STREAM FILE? as in no errors on agi debug, but just doesn't playback the file? It seems to work if I install asterisk 1.0, but 1.0.5 and 1.0.7 break. |
00:26.10 | elriah | No errors at all. Verbose set to 6. agi debug on |
00:26.24 | *** join/#asterisk Thus0 (~Thus0@dyn-83-154-121-98.ppp.tiscali.fr) |
00:26.49 | Nugget | then I have to wonder if the device is reaching the server at all |
00:27.00 | FuriousGeorge | *CLI> Mar 25 19:26:36 NOTICE[824]: chan_sip.c:7278 handle_request: Failed to authenticate user Brian <sip:Brian@10.0.0.2>;tag=4035506318 |
00:27.01 | FuriousGeorge | Mar 25 19:27:04 NOTICE[824]: chan_sip.c:7681 handle_request: Registration from 'Brian <sip:Brian@10.0.0.2>' failed for '10.0.0.100' |
00:27.13 | Nugget | oh, I conflated the two questions, sorry |
00:27.20 | bkw_ | FuriousGeorge, show me your sip.conf entry |
00:27.21 | bkw_ | please |
00:27.25 | bkw_ | put it on pastebin.ca |
00:27.44 | *** join/#asterisk JerJer[mobile] (~jj@65.173.197.174) |
00:29.43 | FuriousGeorge | bkw_: thanks a ton, its there |
00:29.56 | drumkilla | FuriousGeorge: you have to copy the address back to him :) |
00:30.15 | Nugget | http://pastebin.ca/8254 |
00:30.16 | Nugget | most recent post |
00:30.28 | bkw_ | WRONG |
00:30.30 | bkw_ | hold up |
00:30.49 | FuriousGeorge | drumkilla: what you mean? |
00:30.56 | FuriousGeorge | bkw_: be gentle, im new |
00:31.12 | drumkilla | FuriousGeorge: nevermind, Nugget got it |
00:31.26 | FuriousGeorge | ohhhh, the confimation, i got you |
00:31.30 | drumkilla | :) |
00:31.33 | FuriousGeorge | ill facilitate, next time |
00:31.38 | drumkilla | no problem |
00:31.44 | bkw_ | http://pastebin.ca/8255 |
00:32.04 | harryvv | Whats a common cause of me hearing the other party though a voip carrier but the other party not hearing me? I do have DMZ pointed to my server for the iaxport. Did that after it failed the first time and still not make a difference. |
00:32.37 | bkw_ | if you wish to use md5secret you need to do echo -n user:realm:pass | md5sum |
00:32.46 | bkw_ | and put it in as md5secret |
00:32.49 | bkw_ | and not use secret |
00:33.05 | bkw_ | brb |
00:33.15 | FuriousGeorge | i get it, thanks again |
00:33.38 | harryvv | anyway will work on this more latter. |
00:36.24 | bkw_ | drumkilla, was I nice enuf? |
00:37.02 | drumkilla | bkw_: that was great :D |
00:37.36 | bkw_ | I swear i'm tired |
00:37.38 | bkw_ | was up at 4am |
00:37.40 | bkw_ | dog was sick |
00:37.45 | bkw_ | took her to vet at 8:30 |
00:37.45 | drumkilla | :( I'm sowwy |
00:37.49 | bkw_ | she's all better now |
00:37.55 | bkw_ | she goes to the dr. more than I do |
00:38.39 | dogz- | breed? |
00:38.55 | dogz- | btw hi, heh |
00:39.03 | bkw_ | she's a mutt |
00:39.09 | bkw_ | little terrier in her |
00:39.12 | bkw_ | smart as a whip |
00:39.16 | bkw_ | took one day to house train her |
00:39.21 | bkw_ | she had exactly one accident in the house |
00:39.38 | bkw_ | and she won't chew on stuff that isn't hers |
00:40.34 | *** join/#asterisk IQ (~iq@70-59-164-47.omah.qwest.net) |
00:40.54 | bkw_ | g726 will now work with broadvoice btw |
00:40.55 | dogz- | i wish the VP's dog would be like that (refering to chewing on stuff), she thought it be a great boost for company moral if she brought the ~8month old coon hound puppy in |
00:41.11 | dogz- | couple hours later i get a call asking what the blue wire is to, and can it be spliced |
00:41.21 | bkw_ | drumkilla, that rtp fix was needed for that to work.. and it is proper. |
00:41.25 | dogz- | little bugger ate through a cat5 |
00:41.30 | bkw_ | OUCH |
00:41.33 | bkw_ | ZAPPPPP |
00:41.36 | dogz- | rebooted my 2950 |
00:41.48 | FuriousGeorge | bkw_: for whatever reason im still getting a failed to authenticate |
00:41.49 | cripito | bkw_ about the deadlock |
00:42.23 | cripito | i don't have 1.0.7 but have CVS head.. |
00:42.28 | *** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc) |
00:42.37 | bkw_ | cripito, no clue if it locks up on cvs-head |
00:42.42 | bkw_ | I know 1.0.7 and cdr_mysql = FUCKED |
00:42.47 | cripito | no... let me explain |
00:42.51 | cripito | yes is what i mean |
00:42.59 | cripito | before i was running 2 cdr |
00:43.00 | cripito | 3 |
00:43.06 | drumkilla | bkw_: i added the fix |
00:43.13 | bkw_ | drumkilla, for rtp? |
00:43.13 | cripito | cdr_csv, cdr_Mysql and rate_engine |
00:43.15 | bkw_ | I seen it |
00:43.24 | drumkilla | bkw_: I don't think I have ever changed cdr_mysql ... |
00:43.38 | cripito | as soon i take out the cdr_mysql everything works again |
00:43.49 | cripito | but rate_engine is based in mysql too |
00:43.57 | bkw_ | ya i'm not saying is a mysql issue |
00:44.03 | bkw_ | its an issue in the code of cdr_mysql |
00:44.29 | cripito | could be in the implementation of queue? |
00:44.34 | cripito | let me see the code wait a sec |
00:44.43 | bkw_ | no |
00:44.51 | bkw_ | its a deadlock.. or a block |
00:46.44 | AgiNamu | md5secret is transparent to clients right? |
00:46.51 | bkw_ | yes |
00:47.14 | bkw_ | you can't have both |
00:47.15 | AgiNamu | so it'd be possible to implement a SHA512secret if we wanted to. |
00:47.35 | AgiNamu | but the client still sends the user/pass however they normally do it. |
00:47.45 | AgiNamu | it's just an internal Asterisk featuer to keep it as a hash |
00:48.11 | FuriousGeorge | btw_: any idea what else could be causeing the failure to authenticate now that we have removed the md5 auth. |
00:50.53 | tzanger | heh |
00:50.55 | tzanger | Finally, and most importantly, the act of unclasping the belt lets you know you mean business. You might just be going to the toilet or getting ready for sleep, but you mean business. The trousers aren't even unbuttoned and you mean business. |
00:53.16 | *** join/#asterisk jayeola (~jayeola@dsl-80-43-34-188.access.as9105.com) |
00:53.49 | bkw_ | md5secret is just a way to keep the plain text out of the conf file |
00:54.02 | booyeah23 | it wont stop replay attacks |
00:54.16 | bkw_ | actually it does |
00:54.20 | jayeola | hi guys. compiling asterisk, got this err msg "make: *** [asterisk] Error 1; /usr/bin/ld: cannot find -lssl" |
00:54.21 | bkw_ | you need to go read the sip spec |
00:54.29 | bkw_ | install openssl-devel |
00:54.31 | jayeola | what is lssl? |
00:54.34 | FuriousGeorge | i get that, i have a small understanding of checksums. im wondering why my softphone cant authenticate on my server |
00:54.37 | tzanger | out of the conf file? I thought it was to keep the cleartext off the wire |
00:54.43 | bkw_ | nope |
00:54.47 | *** join/#asterisk BuckRogers (~steve@ool-18bce89c.dyn.optonline.net) |
00:54.50 | bkw_ | the user/pass isn't cleartext on the wire right now |
00:54.54 | bkw_ | DUH |
00:55.02 | tzanger | I wasn't aware of that |
00:55.15 | tzanger | I've studied IAX2 a little but never the authentication |
00:55.24 | bkw_ | we are talking sip |
00:55.27 | tzanger | oh |
00:55.28 | BuckRogers | hey i just got a handy tone 486 and it seems like it is not getting a IP i can console in but the wan is not funktioning |
00:55.29 | bkw_ | auth=md5 works on iax |
00:55.32 | tzanger | nevermind then :-) |
00:55.48 | BuckRogers | i can not get it to register |
00:56.16 | BuckRogers | is thier some odd setting for these ata's |
00:56.20 | booyeah23 | i dont think a hashing mechanism with nonces will stop replay attacks |
00:56.25 | BuckRogers | i went through the user manual |
00:57.51 | bkw_ | booyeah23, actually I think it will |
00:58.07 | bkw_ | does it matter |
00:58.17 | bkw_ | most people are dumb enuf to put context=default in general in sip.conf |
00:58.20 | bkw_ | thus ANYONE can make calls |
00:59.04 | *** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc) |
01:02.44 | Nugget | only if default also includes trunkld or trunklocal or equivalent. |
01:03.56 | bkw_ | well some people dont know better |
01:04.21 | Nugget | that cannot be denied. |
01:04.41 | bkw_ | that big SECURITY document doesn't help them |
01:04.57 | bkw_ | like include the outbound context into the inbound menu |
01:05.07 | bkw_ | NOT A WISE THING |
01:06.37 | bkw_ | ok its sad.. but who here is tired of hear about Schiavo |
01:06.49 | Nugget | don't forget to also put voicemailmain(${CALLERID}) in the default context. |
01:07.01 | Nugget | so the whole world can listen to your voicemail |
01:07.01 | bkw_ | hahahahhahahaha |
01:07.11 | bkw_ | ya if you fake the callerid |
01:07.14 | Nugget | right |
01:07.53 | bkw_ | I think a mailboxes settings should live in a .box file or something inside the mailbox |
01:07.58 | bkw_ | such as playback prefs |
01:08.05 | bkw_ | to force asking for a password or not |
01:08.06 | *** join/#asterisk Rob- (~robbie@haylott.plus.com) |
01:08.33 | cripito | the stable version could work with realtime? |
01:08.41 | bkw_ | no |
01:08.44 | bkw_ | stable can't do realtime |
01:08.52 | bkw_ | not without ALOT of patches |
01:08.55 | cripito | :( so is still sip_friends for stable.. |
01:09.00 | bkw_ | go use head |
01:09.06 | bkw_ | strap on a pair and use CVS-HEAD |
01:09.10 | bkw_ | I run it |
01:09.13 | cripito | yeap.. i am using head |
01:11.19 | Nugget | bkw_ loves head |
01:11.37 | file[laptop] | giving it too |
01:11.47 | bkw_ | haha you guys are not right |
01:11.50 | bkw_ | file[laptop], how would you know? |
01:12.37 | Mavvie | anybody with RTP stream insight here? |
01:13.07 | Mavvie | sometimes my peer sends me packets on one port higher: |
01:13.18 | Mavvie | <PROTECTED> |
01:13.18 | Mavvie | <PROTECTED> |
01:13.33 | Mavvie | And I have no idea why that is. |
01:13.41 | bkw_ | depends |
01:13.44 | bkw_ | are you using cvs-head? |
01:13.45 | bkw_ | or stable? |
01:13.59 | Mavvie | My side is 1.0.7, the other is a non-asterisk box |
01:14.08 | bkw_ | ok |
01:14.15 | bkw_ | is asterisk doing register? |
01:14.20 | bkw_ | or is something registering to atserisk? |
01:14.29 | Mavvie | no registration, it's just pushing through calls. |
01:14.40 | bkw_ | nothing doing a register? |
01:14.41 | bkw_ | notify? |
01:14.43 | bkw_ | or anything |
01:14.46 | bkw_ | ok |
01:14.48 | bkw_ | here is what can happen |
01:14.54 | bkw_ | in cvs-stable if its still there |
01:15.04 | bkw_ | even if something comes in we were allocating rtp for it |
01:15.07 | bkw_ | till it died off |
01:15.15 | bkw_ | I think cvs-head fixed it already |
01:15.18 | bkw_ | but I do recall this |
01:15.26 | bkw_ | someone was running out of rtp ports or something |
01:15.35 | cripito | we need bk often here :) |
01:15.46 | cripito | more often i mean |
01:16.18 | cripito | one question.... |
01:16.22 | cripito | in cvs head... |
01:16.40 | cripito | i have a client using firefly 3th party |
01:17.16 | *** join/#asterisk verge (~jfargen@rrcs-67-78-209-206.se.biz.rr.com) |
01:17.50 | cripito | in the last distro when the guy try to loguin asterisk blows up... after try 2 register the firefly probably a 5000 times |
01:17.53 | Mavvie | aha, but the thing is it's coming in from the other side (non-asterisk). |
01:18.10 | bkw_ | Mavvie, so |
01:18.19 | bkw_ | it could be any sip packet that could trigger it |
01:18.23 | bkw_ | notify |
01:18.27 | bkw_ | ANYTHING |
01:18.57 | Mavvie | let me check the traffic again then. |
01:19.04 | bkw_ | cripito, get a backtrace |
01:19.07 | bkw_ | and we can see why |
01:19.56 | *** join/#asterisk seong (~seong@219.94.59.19) |
01:20.20 | BuckRogers | any one experince with grandstream 486 and asterisk |
01:20.32 | cripito | yup |
01:20.37 | BuckRogers | we have a sipura and a softphone already working |
01:20.49 | cripito | as soon the guy comes in again you got it. |
01:20.49 | BuckRogers | but the grand stream config just doesnt want to work |
01:21.08 | Mavvie | funny. I couldn't get the sipura to work, but the grandstream was easy :-) |
01:21.25 | BuckRogers | really ive set all the addresses |
01:21.27 | cripito | mm i think i can simulate it |
01:21.31 | BuckRogers | it doenst even try to registar |
01:21.57 | *** join/#asterisk IQ (~iq@70-59-164-47.omah.qwest.net) |
01:23.05 | BuckRogers | any tips Mavvie |
01:30.47 | *** join/#asterisk macTijn (martijn@linda.net.insecure.nl) |
01:35.12 | *** join/#asterisk lapdr (~inband@ca-fulrtn-cuda2-c2a-243-a.anhmca.adelphia.net) |
01:35.36 | lapdr | hi |
01:36.22 | lapdr | anyone awake? |
01:36.53 | *** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net) |
01:37.16 | *** join/#asterisk znoG (gs@200.115.216.109) |
01:37.40 | voiper | hi guys. i am getting a 404 not found while I try to connect to CISCO 5350. when I do sip debug i see this message does this mean anything ? Answering with non-codec capability 0x1 (telephone-event) |
01:37.50 | newl | lapdr: Nobody here but us chickens. |
01:37.57 | cripito | bk still around? |
01:38.05 | lapdr | ;P |
01:38.27 | cripito | bkw_ still around? |
01:39.11 | lapdr | I'm wanting to get ISDN NT mode support on NI1 to be able to connect digital isdn sets to Asterisk... but there doesn't seam to be much support/interest... so I thought I might see if anyone here has any leads |
01:39.21 | *** join/#asterisk brc__ (~brian@brc.base.supporter.pdpc) |
01:39.25 | cripito | http://www.pastebin.com/262814 even with debug and trace on |
01:42.29 | bkw_ | ok you know you're lame when you have william shatner do a local cable comercial for your law office |
01:42.53 | newl | Denny Crane |
01:43.12 | lapdr | lol |
01:43.33 | Mavvie | denny crane rules :-) |
01:43.40 | Mavvie | too bad he was there for only one episode. |
01:44.55 | newl | Denny is usually always in it. Danny OTOH gets guest appearances. |
01:45.31 | dogz- | Does Shatner do anything besides commercials now? |
01:45.46 | Mavvie | one season, not one episode |
01:49.30 | *** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
01:49.42 | PTG123 | anyone here have a problem undefined symbol ast_pthread_create when compiling asterisk? |
01:49.55 | bkw_ | include utils.h |
01:50.01 | cripito | sounds familiar.. yeap ... |
01:50.02 | bkw_ | or is it util.h |
01:50.06 | tainted- | what does it mean 'unable to find a path from ulaw to 729' |
01:50.14 | cripito | g729 maybe? |
01:50.19 | brc__ | out of licenses |
01:50.36 | tainted- | i have 729 license |
01:50.42 | PTG123 | hmm someone should fix the cvs :) |
01:50.49 | bkw_ | or is it lock.h |
01:50.52 | brc__ | hahah |
01:50.56 | bkw_ | ya ya thats it |
01:50.57 | PTG123 | but including a .h isn't gonna fix it.. its a loading problem, when loading the module |
01:51.12 | tainted- | how do i make sure the g729 module is working? |
01:51.19 | PTG123 | chan_modem.so fails to load |
01:51.19 | bkw_ | PTG123, what modele? |
01:51.21 | bkw_ | er module? |
01:51.25 | brc__ | try g729 show <tab> |
01:51.25 | PTG123 | chan_modem.so |
01:51.36 | bkw_ | loaded fine here |
01:51.42 | PTG123 | ast_pthread_create is defined in the core, so that should def. be loaded before modules right? |
01:51.57 | bkw_ | not if you don't include the headerfile.. but mine works fine |
01:52.12 | PTG123 | just pulled it fresh off of cvs |
01:52.14 | PTG123 | and type gmake |
01:52.16 | PTG123 | and ran it |
01:52.23 | cripito | bkw definitelly cvs head have some problems with the seed and the mapping b/c is also unable 2 read the accountcode from the db |
01:52.39 | bkw_ | you're using freebsd? |
01:52.45 | cripito | me? |
01:52.47 | cripito | no FC3 |
01:52.50 | bkw_ | not yet |
01:52.56 | bkw_ | PTG123, because he says gmake |
01:53.06 | brc__ | eh? |
01:53.12 | bkw_ | cripito, whats broken about it? |
01:53.13 | bkw_ | do tell |
01:53.18 | brc__ | yes |
01:53.30 | PTG123 | yes i am :) |
01:53.33 | bkw_ | do you have an accountcode column in the database |
01:53.37 | cripito | yes |
01:53.41 | bkw_ | PTG123, use supported OS or fix it and submit a patch |
01:53.41 | PTG123 | bkw_: why would i use anything else? :) |
01:53.47 | bkw_ | I don't like linux |
01:53.51 | bkw_ | but i'm not gonna pull my hair out |
01:53.59 | cripito | :D |
01:54.00 | bkw_ | with a half ass supported os for asterisk |
01:54.02 | PTG123 | bkw_: every version has worked fine on bsd up to now |
01:54.05 | tainted- | i'm trying to get my grandstream 486 to talk to asterisk using g729 |
01:54.07 | PTG123 | bkw_ supported it great |
01:54.18 | tainted- | but it keeps saying NOTICE[13195]: channel.c:1724 ast_set_read_format: Unable to find a path from g729 to ulaw |
01:54.30 | brc__ | tainted-, did you try the command I suggested? |
01:54.32 | brc__ | what did it say? |
01:54.39 | tainted- | it's not a command |
01:54.43 | bkw_ | lots show translation |
01:54.46 | tainted- | g729 show <tab> |
01:54.47 | bkw_ | s/lots/ |
01:54.50 | brc__ | leme check |
01:55.27 | tainted- | g729 - - - - ... |
01:55.34 | bkw_ | load the codec boi |
01:55.37 | tainted- | i'm assuming it's not doing translation correctly |
01:55.44 | bkw_ | you don't have a codec |
01:55.45 | bkw_ | duh |
01:55.47 | tainted- | bkw_ in modules.conf? |
01:55.54 | tainted- | i do |
01:55.55 | brc__ | you don't have g729 loaded correctly |
01:55.57 | file[laptop] | the codec fails to be loaded |
01:56.04 | cripito | try load codec_g729a.so |
01:56.05 | bkw_ | really? |
01:56.06 | MikeJ[Jayden] | it's fat albert.... |
01:56.08 | MikeJ[Jayden] | hey hey hey |
01:56.26 | PTG123 | what port is ident on> |
01:56.40 | bkw_ | 113 |
01:56.47 | brc__ | ident is lame |
01:56.47 | bkw_ | aka auth |
01:56.59 | bkw_ | brc_ is lame |
01:57.05 | *** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net) |
01:57.07 | brc_ | asterisk is lame |
01:57.11 | tainted- | ok that worked |
01:57.16 | tainted- | i'm an dumb ass |
01:57.26 | file[laptop] | brc_ is gay! |
01:57.34 | tainted- | i moved the codec.so and forgot about it |
01:57.41 | MikeJ[Jayden] | file[laptop]... hehe |
01:57.44 | file[laptop] | ooh baby touch me down there |
01:57.44 | tainted- | but i still get Mar 25 17:56:59 WARNING[13195]: dsp.c:1469 ast_dsp_process: Inband DTMF is not supported on codec g729. Use RFC2833 |
01:57.52 | tainted- | when i have dtmfmode = rfc2833 |
01:58.00 | tainted- | both in sip.conf and on the ATA |
01:58.15 | bkw_ | must not |
01:58.23 | cripito | :) check your peer definition |
01:58.29 | cripito | b/c definitelly looks like not |
01:58.50 | bkw_ | your conf file is tainted |
01:58.50 | brc_ | whooop whooooooop, warning, configuration error detected |
01:58.51 | bkw_ | :P |
01:58.51 | cripito | ok bkw_ coming back to the point i have 2 versions of cvs head |
01:59.07 | bkw_ | k |
01:59.07 | tainted- | problem is.. peer is ulaw |
01:59.09 | bkw_ | and? |
01:59.09 | cripito | 1 works fine |
01:59.23 | bkw_ | diff them |
01:59.31 | file[laptop] | bkw_: we need more Mac hardware |
01:59.36 | bkw_ | file yes I know |
01:59.38 | tainted- | and peer<---ulaw--->*<---g729--->gs486ATA |
01:59.40 | bkw_ | I want MORE MORE MORE |
01:59.51 | file[laptop] | bkw_: wanna go half on an Xserve and we can colo it? :p |
01:59.54 | brc_ | file[laptop], you are utterly corrupted |
01:59.57 | bkw_ | file tempted |
02:00.01 | brc_ | I'll go 1/3 |
02:00.08 | bkw_ | I have colo space |
02:00.17 | file[laptop] | eep Peacekeeper Wars |
02:00.18 | file[laptop] | bbl |
02:00.23 | Nugget | yay xserve |
02:00.28 | bkw_ | once ya go mac you never go back |
02:00.35 | cripito | :D i have a spare server |
02:00.36 | Nugget | yup |
02:00.45 | bkw_ | I went mac.. never gonna go back |
02:00.45 | cripito | Asterisk CVS-HEAD-02/16/05-13:50:59 built by root@sls-ce2p4 on a i686 running Linux |
02:00.45 | cripito | <PROTECTED> |
02:01.21 | cripito | #define BUILD_VERSION "CVS-HEAD-03/25/05-20:22:08" this one |
02:01.21 | bkw_ | cripito, step forward day by day till it breaks from 2/15 |
02:01.26 | bkw_ | er 2/16 |
02:01.30 | cripito | wow!! |
02:01.31 | cripito | ok |
02:01.31 | bkw_ | or jump 4 days at a time |
02:01.38 | bkw_ | then zero in on when it broke |
02:01.46 | cripito | that's sounds better |
02:01.47 | bkw_ | thats usually what I do when I can't find it |
02:02.03 | MikeJ[Jayden] | once you get to day, get to the specific file version that broke it if there are multiple that day |
02:02.05 | bkw_ | its amazing when something totally unrelated chances.. then breaks something else |
02:02.09 | *** join/#asterisk jayeola (~jayeola@dsl-80-43-34-188.access.as9105.com) |
02:02.17 | cripito | yeap i known :) |
02:02.32 | cripito | ok bk i will |
02:03.28 | tainted- | is rfc2833 supposed to be ass? |
02:03.40 | bkw_ | no |
02:03.41 | bkw_ | it works great |
02:03.48 | MikeJ[Jayden] | sip is ass |
02:03.49 | tainted- | keeps missing digits |
02:03.53 | bkw_ | no sip isn't ass |
02:03.56 | bkw_ | just the way asterisk does it is |
02:03.57 | MikeJ[Jayden] | :) |
02:04.18 | MikeJ[Jayden] | ok, how about sip is very complicated for just voice... is that better :) |
02:04.38 | bkw_ | what dual 2.5ghz with 1.25 FSB not good enuf? |
02:04.39 | jayeola | i keep getting this error "/usr/bin/ld: cannot find -lssl " even though i have libidn libidn-devel installled |
02:04.49 | bkw_ | dis I say libidn |
02:04.49 | cripito | taitend 1 question |
02:04.51 | bkw_ | you dummy |
02:04.54 | bkw_ | I said OpenSSL |
02:04.57 | bkw_ | OpenSSL-Devel |
02:04.57 | Nugget | I'd kick myself if I bought one and a month later the new ones came out. |
02:05.00 | cripito | your provider is? |
02:05.02 | Nugget | and new ones are overdue |
02:05.14 | bkw_ | -lssl = OpenSSL |
02:05.29 | jayeola | aha! |
02:05.36 | bkw_ | Nugget, honestly this 1.6ghz G5 is whippin the piss out of my P4 |
02:06.08 | bkw_ | 1.2ghz G4 in my ibook |
02:06.08 | Nugget | I believe it |
02:06.14 | *** join/#asterisk infra (~infra@216-251-177-106.ips.cpinternet.com) |
02:06.15 | bkw_ | 1.6ghz G5 in my imac |
02:06.21 | bkw_ | 768 in my ibook |
02:06.25 | Nugget | I'm on a 1ghz powerbook right now. coming up on three years old and it's time to move on. |
02:06.26 | bkw_ | 1 gig ram in my imac |
02:06.57 | infra | hi, anyone with Quicknet PhoneJack experience? |
02:07.08 | cripito | :) |
02:07.09 | bkw_ | RUN FOR THE HILLZ |
02:07.13 | cripito | asterisk in mac? |
02:07.18 | cripito | good 2 known |
02:07.28 | bkw_ | I run asterisk on my ibook |
02:07.30 | MikeJ[Jayden] | asterisk runs everywhere man..... |
02:07.30 | bkw_ | works great |
02:07.35 | Nugget | I run asterisk on my powerbook |
02:07.40 | bkw_ | its fun |
02:07.41 | tainted- | what is DTMF Payload Type and what should it be? |
02:07.46 | bkw_ | 101 |
02:07.48 | bkw_ | AVT |
02:08.16 | cripito | mike :) now yes... was funny find the win32 version for moco$ |
02:08.33 | bkw_ | and did he put up patches yet |
02:08.34 | MikeJ[Jayden] | I have had it on windows for a few months now |
02:08.53 | MikeJ[Jayden] | bkw_, I e-mailed with him, he said monday |
02:09.19 | cripito | i prefer the linux version... |
02:09.22 | MikeJ[Jayden] | but his runs a lot smoother than my hack job... so I really want to see his patches |
02:09.28 | jayeola | open-ssl ain't in my repositary, where can i get it from? sourceforge? |
02:09.33 | bkw_ | AsteriskWin32 Source Code 0.51 |
02:09.33 | bkw_ | You need asterisk 1.0.5 source code at ftp.asterisk.org |
02:09.33 | bkw_ | and AsteriskWin32 cygwin patches, avaible soon, come back later ! |
02:09.40 | bkw_ | Mike I doubt it |
02:09.47 | bkw_ | I so totally doubt it |
02:09.58 | bkw_ | ya the "working" linux is the best |
02:10.02 | MikeJ[Jayden] | I know, I they are not out there next week.... |
02:10.06 | MikeJ[Jayden] | but we will see |
02:10.11 | cripito | jayeola what SO are you running? |
02:10.48 | cripito | ok any way to bring the daily patch for asterisk apart of cvs? |
02:11.03 | MikeJ[Jayden] | cripito, huh? |
02:11.09 | hmodes | which do you seek? |
02:11.21 | bkw_ | OMG ITS HMODES |
02:11.23 | cripito | all from 2/16 |
02:11.24 | bkw_ | hey honey how are you? |
02:11.33 | cripito | till today |
02:11.33 | hmodes | howdy bkw :) |
02:11.39 | bkw_ | cvs co -d "2005/02/16" asterisk |
02:11.40 | hmodes | i'm ok i guess, you? |
02:11.41 | bkw_ | or some shit like that |
02:11.43 | bkw_ | man cvs |
02:11.43 | cripito | yea yeap |
02:11.56 | bkw_ | oh don't forget the -r |
02:11.57 | cripito | i have 2 pull a months!!! don't be bad guy ;) |
02:12.01 | bkw_ | for v1.0 |
02:12.02 | bkw_ | or what ever |
02:12.16 | bkw_ | ;) |
02:12.18 | jayeola | cripito: i'm a bit s-l-o-w. all i had to do was look for openssl and not "open-ssl" |
02:12.34 | cripito | yes jayeola.. all that u need is openssl |
02:12.39 | cripito | devel |
02:12.44 | jayeola | uh-huh |
02:12.50 | cripito | if u have yum |
02:12.52 | jayeola | <-- apt-getting |
02:12.57 | cripito | better |
02:13.06 | infra | Seeking Quicknet experience; have rxgain problem. |
02:13.14 | cripito | u need openssl and openssl-devel |
02:13.35 | cripito | ok bkw i am bringing the patches i will tell u what day broke |
02:19.07 | *** join/#asterisk abcdefgh (~infra@216-251-177-106.ips.cpinternet.com) |
02:22.32 | infra | Anyone have experience with Quicknet Internet PhoneJack cards? |
02:23.07 | newl | When the A party calls B party and their number is not presenting, is there a common string that's given such as "private number" or does Asterisk mung that for common output? |
02:23.10 | infra | also...anyone recommend best current apps reference (old *docs book is from 2003)? |
02:25.20 | bkw_ | app_skel.c |
02:26.12 | Damin | What about app_moospenis? |
02:27.34 | bkw_ | ya ya |
02:27.37 | bkw_ | i'm in da conf |
02:27.41 | bkw_ | but nobody is there with me |
02:29.00 | booyeah23 | is it possible for a context to be null? |
02:29.19 | bkw_ | depends.. in some cases it falls back to default |
02:29.21 | booyeah23 | what does it mean context ' ' |
02:29.30 | bkw_ | paste the message in here |
02:29.34 | booyeah23 | i think the problem im having with this bridging thing is because of it |
02:30.05 | booyeah23 | <PROTECTED> |
02:30.15 | booyeah23 | ignore that |
02:30.16 | booyeah23 | Unable to find extension '12' in context '' |
02:30.23 | booyeah23 | basically im bridging to calls together |
02:30.49 | booyeah23 | user1 calls pbx, forwards to user2 |
02:30.52 | booyeah23 | with tT |
02:30.58 | booyeah23 | T doesnt work |
02:31.15 | booyeah23 | t works, user1 can transfer |
02:31.45 | booyeah23 | user2 cannont transfer |
02:31.57 | booyeah23 | *cannot |
02:32.09 | booyeah23 | i fixed the problem with the native bridging |
02:32.50 | booyeah23 | notransfer=yes was not working correctly, so i added a check to just not native bridge if that was set |
02:34.00 | booyeah23 | in chan_iax2.c |
02:34.14 | PTG123 | app_moosepenis is bkw_s favorite app |
02:34.38 | booyeah23 | http://pastebin.ca/8257 |
02:35.18 | booyeah23 | i guess i am going to run a debugger on it when it gets the # |
02:35.21 | booyeah23 | and see what the structures are |
02:35.34 | booyeah23 | im guessing the context for the person being called is null |
02:36.31 | jayeola | hurrah! asterisk compiled |
02:36.58 | jayeola | now to get it working with gnomemeeting or sflphone |
02:37.01 | *** part/#asterisk JerJer[mobile] (~jj@65.173.197.174) |
02:42.52 | bparker | is there a trick to getting moh working |
02:44.40 | infra | Is this channel's traffic archived anywhere? |
02:48.10 | tzafrir_laptop | gnomemeeting requires h323, right? |
02:49.37 | jayeola | infra: if you can log this convo in irssi or xchat quite easily |
02:50.05 | jayeola | depends on what irc client you are running |
02:50.14 | FuriousGeorge | can someone give me a hand getting x-lite aquthenticated with my very preliminary configuration of asterisk |
02:50.15 | *** join/#asterisk Othello (Othello@nusnet-219-211.dynip.nus.edu.sg) |
02:50.27 | FuriousGeorge | cuz im at a wall over here |
02:50.30 | *** join/#asterisk sedwards50 (~chatzilla@adsl-66-120-116-250.dsl.sndg02.pacbell.net) |
02:50.46 | infra | I mean a searchable archive of old traffic... |
02:51.05 | infra | Perhaps my problems have been discussed... |
02:51.06 | sedwards50 | anybody have experience with info-digits/ani2? |
02:51.24 | cripito | guys anyone can recommend a phone for linux? |
02:51.34 | FuriousGeorge | softphone? |
02:51.47 | Syncros | iaxcomm |
02:51.49 | Shido6 | FuriousGeorge - user and peer |
02:51.54 | Shido6 | brb - Pistons Game |
02:52.11 | cripito | yes softphone |
02:52.47 | sedwards50 | iaxcomm 1.0rc2 seems a bit buggy |
02:52.52 | FuriousGeorge | Shido6: ill try that, but the exaples i see list it as friend always |
02:53.29 | FuriousGeorge | cripto: let me preface this by saying ive never used any. there's also gnuphone, and if u contact xten u can get a beta of xten for linux |
02:53.52 | sedwards50 | Does anybody know how to read the info digits (aka ani2) from a pri t1? |
02:53.56 | jayeola | anyone tried/used this; http://sflphone.org ? |
02:53.56 | cripito | mmmmm that one sounds nices |
02:54.08 | bparker | does anyone know what codec moh uses by default |
02:54.50 | cripito | sip... mmm lets trying.. thanks guys |
02:55.23 | infra | In the absence of any experience on Quicknet, anyone recommend a hardware line amp? |
02:57.07 | *** join/#asterisk dersteer (~travis@24.231.151.119.gha.mi.chartermi.net) |
02:57.50 | FuriousGeorge | shido6: i tried that. x-lite says "404 Not Found" and asterisk says: Mar 25 21:57:32 NOTICE[4901]: chan_sip.c:7681 handle_request: Registration from 'brian <sip:brian@10.0.0.2>' failed for '10.0.0.100' |
02:57.55 | infra | I guess I'll try during business hours :-) |
02:58.46 | newl | FuriousGeorge: I've got mine configured as friend and it works fine. |
02:59.14 | FuriousGeorge | newl: yeah i tried that too |
02:59.29 | FuriousGeorge | keep in mind im very new to this. |
03:00.00 | FuriousGeorge | besides the obvious parts where i define my sip clients, give them pw's etc, what else is required for x-lite to log into * |
03:00.08 | jayeola | any uk users here? or from euroland? i'm looking for a good sip provider |
03:01.37 | *** join/#asterisk clh (~clh@adsl-69-107-18-183.dsl.pltn13.pacbell.net) |
03:02.20 | *** join/#asterisk sedwards50 (~chatzilla@adsl-66-120-116-250.dsl.sndg02.pacbell.net) |
03:02.32 | *** part/#asterisk clh (~clh@adsl-69-107-18-183.dsl.pltn13.pacbell.net) |
03:03.03 | *** join/#asterisk doughecka (~Doug@doughecka.user) |
03:03.40 | newl | FuriousGeorge: Try here and see how you go. http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20xten%20xlite |
03:04.27 | jayeola | thanks |
03:05.05 | jayeola | <-- just compiled asterisk successfully. using google and this guide; http://www.automated.it/guidetoasterisk.htm#_Toc49248757 |
03:08.44 | FuriousGeorge | newl: thanks, that was the guide i found and used earlier today, and i had forgotten about it. i still get the same error but maybe after i apply the registry changes it recommends... |
03:09.49 | newl | FuriousGeorge: maybe. I've not tried it from the Win32 client. The Linux version works as expected though. |
03:10.39 | Othello | hi guys |
03:11.05 | Othello | I got a little problem ... |
03:11.34 | Othello | it seems that the console driver for asterisk will NOT work on a ES1371 sound card on a real machine |
03:11.54 | Othello | I have tried it on a VMWare machine and it woerks |
03:11.58 | Othello | :/ |
03:12.23 | niZon | The console driver doesn't seem to work in vmware for me |
03:12.26 | niZon | unless i missed something |
03:12.37 | jayeola | newl: did you use some kinda phone to get it working or regular headphones and a mic? |
03:12.45 | Othello | niZon: what kernel are you running? |
03:12.58 | Othello | and niZon: which version of VMWare? |
03:13.05 | niZon | 2.4.27 |
03:13.05 | newl | jaiger: headphone/mic |
03:13.12 | niZon | vmware umm |
03:13.20 | niZon | 4.5.2 |
03:14.07 | Othello | ah ok |
03:14.13 | Othello | you got the right audio drivers working? |
03:14.22 | niZon | probablly not :P |
03:14.44 | Othello | it's supposed to emulate an es1371 |
03:14.50 | Othello | go configure that in your kernel |
03:14.57 | niZon | mk |
03:15.02 | niZon | does ztdummy work for you? |
03:15.10 | Othello | or try 'modprobe es1371' |
03:15.37 | Othello | niZon ... sorry for asking but what does ztdummy do? |
03:15.50 | niZon | fake zaptel device for timing |
03:15.56 | niZon | used by meetme |
03:16.43 | Othello | I see... |
03:16.49 | Othello | :S ... |
03:16.59 | Othello | I still have a lot to learn |
03:18.28 | niZon | yeah me too |
03:18.44 | niZon | what does your dialplan look like for dialing console? |
03:18.59 | *** join/#asterisk klasstek (~nunyobiz@c-24-9-148-246.client.comcast.net) |
03:19.06 | newl | phone $HOME |
03:19.53 | bkw_ | so guys |
03:20.26 | file[laptop] | yes bkw? |
03:21.46 | *** join/#asterisk jayeola (~jayeola@dsl-80-43-34-188.access.as9105.com) |
03:21.51 | Qwell | hmm |
03:21.53 | Shido6 | you can use it for iax2 trunking |
03:21.54 | Shido6 | too |
03:22.34 | Othello | niZon ... |
03:22.34 | Qwell | Shido6: Do I have to call to get that "other" DID removed from my account? |
03:22.34 | Othello | I don't have a dialplan |
03:22.34 | Shido6 | did removed? |
03:22.34 | Othello | I used the "make samples" thingy |
03:22.36 | Shido6 | is it a MI number? |
03:22.40 | Othello | and just type 'dial' on the console |
03:22.43 | Qwell | no, tollfree. the one that was mapped...somewhere |
03:22.44 | Shido6 | dont make samples |
03:22.57 | Shido6 | tollfree? hrmm |
03:23.00 | Shido6 | dont worry about it |
03:23.08 | niZon | Othello: um, oh :P |
03:23.10 | Shido6 | ur not getting billed if no one uses it |
03:23.16 | Qwell | ok |
03:26.53 | FuriousGeorge | grrr... still nothing. wtf |
03:27.18 | Othello | interestingly enough ... mpg123 works and I do hear a "beep" when I type 'dial' but nothing more |
03:27.38 | Othello | it's almost as if something made the playback thread get stuck |
03:28.29 | Othello | Shido6 ... what do I do then? |
03:28.30 | FuriousGeorge | when i put in the wrong pw it says the same thing as when i put in the right one. what gives. |
03:28.55 | *** join/#asterisk NewSole (~david@i216-58-44-245.avalonworks.net) |
03:29.05 | IQ | Hi, anyone using Festival with * ? |
03:29.49 | *** join/#asterisk sergey (~none@195.151.15.13) |
03:30.01 | *** join/#asterisk jskcr (~jskcr@jskcr.user) |
03:32.01 | jayeola | do i need a -statis- ip? at the moment i have a dynamic one provide, (i think), by my adsl ISP |
03:32.55 | IQ | Festival TTS wit *. Anyone using it ? |
03:33.28 | newl | FuriousGeorge: try setting nat=yes for that entry and see if it still happens. |
03:35.16 | FuriousGeorge | still happens |
03:35.34 | ariel_ | jayeola, you can use services like dyndns.org to setup your ip address and I use my asterisk system with it. |
03:37.12 | FuriousGeorge | what does it mean "handel_request" does it have a problem with my suername or soemthing? |
03:37.38 | ariel_ | FuriousGeorge, your xlite in on the same network as the asterisk is it not? |
03:38.20 | FuriousGeorge | yes |
03:38.37 | Shido6 | . |
03:39.03 | ariel_ | FuriousGeorge, your settings for sip.conf your account is brian? [brain] |
03:39.34 | jayeola | thanks ariel_ ! |
03:40.19 | *** join/#asterisk yaboo (~jsirucka@220.245.131.131) |
03:40.25 | FuriousGeorge | yeah, in parenthesis above the settings for the client |
03:41.26 | ariel_ | that is your login name. what are the settings you have in the xlite? |
03:43.48 | FuriousGeorge | i thought username= sets the username and the thing in parenthesis was just an arbitrary name for the settingfs in that part |
03:44.06 | anachron | when you checkout the latest zaptel via svn .. where does it put the files? |
03:44.11 | FuriousGeorge | cuz in the [] its capital and in the username= and xlite its lowercase |
03:44.14 | ariel_ | In setting the xlite you need username, and authorization user, same brian, your password next and put the asterisk ip address in the domain, and sip proxy only. |
03:44.26 | ariel_ | FuriousGeorge, change it to lower case |
03:44.44 | FuriousGeorge | in process |
03:46.46 | FuriousGeorge | brb |
03:48.27 | *** join/#asterisk Newbie___ (some@60.48.163.164) |
03:49.08 | Newbie___ | hi all, i am about to purchase a Zhone Channel Banks 16FXS & 8FXO, any advice ? |
03:50.23 | ariel_ | Newbie___, good luck. I have had nothing good about the zhones I used. I have 8 of them sitting getting dust in there boxes. I don't think there even good enough to sell on ebay. |
03:51.05 | ManxPower | I assume everyone has seen this? http://www.packet8.net/about/virtual_office.asp |
03:51.08 | *** join/#asterisk Negrobird (~Rober@resnet240001.utdallas.edu) |
03:51.17 | ariel_ | no I have not |
03:52.04 | jayeola | brb |
03:53.03 | *** part/#asterisk sergey (~none@195.151.15.13) |
03:53.46 | Newbie___ | ariel_: is a piece of junk ? |
03:54.32 | ariel_ | Newbie___, the ones I have are. But that just my view of them. I have 8 of them as paper holders. |
03:54.46 | IQ | Newbie___: No, ariel_ is not a piece of junk |
03:55.14 | ariel_ | IQ you want to buy mine 8 c/b |
03:55.49 | Newbie___ | voip-info.org says is good if use as FXS |
03:55.50 | ManxPower | As long as you FULLY UNDERSTAND the limitations of them. |
03:55.52 | Newbie___ | hmmm |
03:55.54 | *** part/#asterisk trig_hm (~jb@home.monkeypr0n.org) |
03:56.06 | IQ | ariel_: I am on your side dude. I'm just telling him that ariel_ is not piece of junk :) |
03:56.12 | NewSole | what are they |
03:56.17 | NewSole | Zhone Channel Banks |
03:56.42 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
03:56.58 | dmccollum | Hello Everyone! |
03:57.04 | ManxPower | ~docs |
03:57.05 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
03:57.05 | Newbie___ | NewSole: the one i am looking at has 16 FXS and 8 FXO |
03:57.07 | ManxPower | ~mailinglist |
03:57.08 | jbot | i guess mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
03:58.38 | ManxPower | I think the Packet8 Virtual PBX a tad expensive. |
03:58.48 | dmccollum | I saw a post in the mailinglist archives about changing the default voice that festival uses and I can't seem to figure out the post from bkw on how it actually works. Anyone care to help me if I post the short post? |
03:59.10 | ManxPower | dmccollum: there are several docs out there, check google |
03:59.17 | ManxPower | ~google festival mbrola |
03:59.47 | IQ | When Digium will go public? |
04:00.02 | ariel_ | Newbie___, hope they work for you. What is the unit going to cost you? |
04:01.19 | dogz- | im currently having problems with * answering incoming calls, outgoing calls work fine.. Im concentrating my efforts in extensions.conf |
04:01.26 | dogz- | is this correct? or should i be looking somewhere else |
04:01.37 | ManxPower | dogz-: Either that or zapata.conf |
04:01.44 | ManxPower | /etc/asterisk/zapata.conf |
04:01.55 | dogz- | thanks |
04:02.17 | ManxPower | dogz-: Do you see ANYTHING in the CLI when a call comes in? |
04:02.24 | dogz- | no |
04:02.33 | ariel_ | argh all they have on the news is this thing with Terry Schiavo |
04:02.36 | Newbie___ | ariel_: 99.00 |
04:02.59 | dogz- | ManxPower: im leaning towards my "incoming" statement, but i may be wrong |
04:03.00 | ariel_ | Newbie___, ebay? |
04:03.09 | Newbie___ | ariel_: yes, but undecided |
04:03.13 | dogz- | i will play with it a bit, see if i cant get this bugger worken |
04:03.31 | dmccollum | Would you rather listen to news about Michael Jackson? |
04:03.45 | Newbie___ | Packet8 offers international plan at 49.90, but cell calls not included |
04:03.46 | ariel_ | dmccollum, no |
04:04.10 | Newbie___ | why dont they include cell calls |
04:04.14 | ariel_ | Newbie___, if you can't return it. think if what your time is worth. |
04:04.37 | ManxPower | dogz-: dunno |
04:05.12 | ManxPower | Newbie___: because in places other than the USA/Canada calling a cell phone costs more than calling landlines. |
04:05.15 | Newbie___ | ariel_: maybe i would go for something more reputable like ADT or ADC i forgot the name |
04:05.19 | ariel_ | Newbie___, international cell calls are more per minute then what they are like in the US and Canada. |
04:05.24 | Qwell | ManxPower: I think I saw a place where it was less |
04:05.31 | ariel_ | Adtran,,,,adtran |
04:06.24 | Newbie___ | there is this web site www.telextreme.com, they include selected countires cell. when i tried to call them, i think the lady know nothing |
04:06.32 | *** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc) |
04:06.42 | Newbie___ | is a MLM company |
04:07.07 | dmccollum | Hmm, Swedish Female MBrola voices. I may have to put the weather extension on my speed dial. |
04:07.55 | FuriousGeorge | ariel: im not sure what i did, but i had some degree of success. now asterisk says -- Registered SIP 'brian' at 10.0.0.100 port 5060 expires 1800 |
04:07.55 | FuriousGeorge | <PROTECTED> |
04:08.14 | *** join/#asterisk angler (~angler@angler.digium.sponsor.pdpc) |
04:08.29 | FuriousGeorge | but when i dial out x-lite makes the busy sound and displays 404: not found |
04:09.33 | ariel_ | FuriousGeorge, then your context is incorrect for your calls |
04:09.38 | *** join/#asterisk jayeola (~jayeola@dsl-80-43-16-212.access.as9105.com) |
04:09.38 | IQ | FuriousGeorge: Once I was getting 404, kept looking on my configuration. but didn't find anything - then I found out that the number I'm dial is busy |
04:10.31 | IQ | FuriousGeorge: Then everything started working - I'm not sure if this could be the case - or maybe I was facing some network issue |
04:10.46 | *** part/#asterisk jayeola (~jayeola@dsl-80-43-16-212.access.as9105.com) |
04:11.22 | FuriousGeorge | ariel: you mean at the top of my sip.conf where i have context=default (for now)? |
04:11.36 | FuriousGeorge | IQ: the number im trying to dial is the info # for the SIP provider |
04:11.58 | Qwell | FuriousGeorge: Do you have a context specified for your xlite user? |
04:12.02 | Newbie___ | life in communication really sucks |
04:12.07 | IQ | FuriousGeorge: I see... well, just wanted to tell you what I experienced |
04:12.13 | FuriousGeorge | maybe it is busy, ill try another. |
04:12.22 | FuriousGeorge | i do appreciate |
04:12.43 | Katty | hi lads |
04:12.54 | IQ | FuriousGeorge: by any chance are you using brain net ? |
04:13.13 | ariel_ | evening Katty |
04:13.20 | FuriousGeorge | i will put context=default with the clients settings, |
04:13.29 | Katty | ariel_: how's teh family? |
04:13.43 | FuriousGeorge | but to be honest, i just picked this up last week so my understanding is weak to say the least |
04:13.50 | *** join/#asterisk jayeola (~jayeola@dsl-80-43-16-212.access.as9105.com) |
04:13.54 | ariel_ | FuriousGeorge, you should not use default for anything other then a trap. |
04:14.11 | FuriousGeorge | trap? |
04:14.30 | ariel_ | Katty, fine thanks. Getting ready for my little girls bday. She will be 2 years old. |
04:14.31 | Qwell | default should "do" anything |
04:14.50 | Katty | ariel_: yay! |
04:14.58 | Katty | ariel_: remember napkins! |
04:15.06 | ariel_ | default should only have exten => X.,1,Congestion |
04:15.24 | Katty | ariel_: i've been working the arms out. they're getting dreamy |
04:15.32 | Katty | ariel_: wanna see? (= |
04:15.42 | ariel_ | sure |
04:15.50 | *** join/#asterisk Grooby (~Grooby@ip24-250-126-171.dc.dc.cox.net) |
04:15.52 | Katty | http://www.brick.net/~izaah/tehflex.jpg |
04:16.04 | yaboo | whats a fwd test number I can call |
04:16.14 | ariel_ | k |
04:16.14 | Qwell | yaboo: 612 I believe |
04:16.41 | yaboo | thanks |
04:16.51 | yaboo | Qwell still trying to sort out my fwd dialing |
04:17.08 | ariel_ | 65342 |
04:17.40 | CoolAcid | My FWD number: 510016 |
04:17.41 | Katty | i think i'm going to try for the croft look |
04:18.03 | Katty | maybe that way i could lift a monitor without help ;) |
04:18.22 | ariel_ | Katty, you look fine the way you are. |
04:18.39 | Katty | ariel_: i'm not toned though. i like toned. |
04:18.58 | FuriousGeorge | x is acting up. brb |
04:19.00 | CoolAcid | Anyone got IAXTel working ok? |
04:19.09 | Qwell | CoolAcid: yep |
04:19.31 | CoolAcid | QWell: Herm.. Mine doesn't dial well, and when it does, it's garbled. But FWD is fine. |
04:20.38 | CoolAcid | Have good ping times to iaxtel.com too |
04:20.55 | ariel_ | CoolAcid, yes |
04:21.12 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
04:21.44 | Newbie___ | ariel_: if i want to connect my cell to *, i would need a GSM destop phone and a FXO , right ? |
04:21.57 | yaboo | damm this fwd is going to tear me around |
04:22.14 | yaboo | followed the voip-info.org page to no success |
04:22.15 | ariel_ | Newbie___, I have no idea. I have never looked into connecting a cell to asterisk |
04:22.42 | *** join/#asterisk Wi_Fi (~OUT@c-24-127-12-85.hsd1.ca.comcast.net) |
04:22.47 | Katty | Newbie___: you sure have a long tail. |
04:23.01 | Newbie___ | it seems logical beacuse GSM desktop phone has a dial tone , but than i can be wrong |
04:23.16 | Newbie___ | Katty: lol |
04:23.39 | yaboo | anyone able to walk me with why fwd under dosen't work |
04:23.45 | CoolAcid | yahoo msg me |
04:25.31 | *** join/#asterisk JunK-Y (~junky@modemcable174.107-81-70.mc.videotron.ca) |
04:26.31 | *** join/#asterisk implicit (~implicit@ip68-7-149-247.sd.sd.cox.net) |
04:30.19 | *** join/#asterisk nix000 (~nix000@66.11.165.71) |
04:30.21 | ariel_ | it's late and I need to get up early. See you all have fun. |
04:31.28 | nix000 | stevek: ping |
04:32.20 | dmccollum | So do I need to download the mbrola binaries also, or just the additional voice files? |
04:34.25 | *** join/#asterisk wildcard` (~blah@d209-121-36-100.bchsia.telus.net) |
04:34.29 | wildcard` | hi every one |
04:34.44 | wildcard` | im new to asterisk im about to be setting up some units inside a few businesses |
04:34.53 | wildcard` | and i was wondering if theirs any access to jamica |
04:34.59 | wildcard` | jamacia |
04:35.04 | implicit | yes |
04:35.15 | wildcard` | awesome ! how many nodes ? |
04:35.19 | wildcard` | or can u tell |
04:35.45 | wildcard` | im gonna be purchasing 2 of these Grandstream Budgetone 100-101 (black) |
04:35.51 | wildcard` | from pulver innovations |
04:35.56 | implicit | what kind of 'access' are you talking about |
04:36.10 | implicit | you want jamaican routes? |
04:36.12 | wildcard` | can i setup a asterisk server here in british columbia canada |
04:36.12 | *** join/#asterisk JT19 (~JT@71.98.88.202) |
04:36.19 | wildcard` | and connect to a land line in jamaica |
04:36.25 | implicit | sure |
04:36.33 | wildcard` | ya jamaican routes |
04:36.40 | wildcard` | is there a list available |
04:36.41 | implicit | what kind of capacity are you looking for |
04:36.42 | wildcard` | of the routes |
04:36.47 | wildcard` | just 1 line |
04:36.50 | implicit | E1, 10xE1, DS3 |
04:36.50 | implicit | oh |
04:37.00 | implicit | you want DIDs or just outbound? |
04:37.26 | wildcard` | i wana have a customer be able to place a call outbound from british columbia over the asterisk network to jamacia |
04:37.35 | wildcard` | if thats how u spell it |
04:37.38 | implicit | asterisk is not a network |
04:37.45 | wildcard` | oh ya thats the PBX |
04:37.47 | implicit | but it is possible to get jamaican routes |
04:37.49 | wildcard` | software rite |
04:37.49 | Shido6 | heh |
04:37.56 | wildcard` | whats that online network |
04:37.57 | dogz- | someone mind taking a quick look at my configs http://pastebin.ca/8262 Im having issues accepting incoming calls... Im thinking its my extension.conf |
04:37.59 | wildcard` | u can join with |
04:38.13 | wildcard` | where you provide your phone |
04:38.21 | wildcard` | in exchange for access to other ppls phones |
04:39.04 | Shido6 | accepting incoming calls from here, dogz and to hat |
04:40.01 | dogz- | ? |
04:41.06 | IQ | HI, how do I call console from an extension? |
04:41.17 | JunK-Y | IQ: huH? |
04:41.27 | JunK-Y | a system command ya mean? |
04:41.34 | JunK-Y | System(blah) |
04:42.20 | IQ | like in dial plan I can set up an extension like 999? |
04:42.31 | CoolAcid | Dial(Console/dsp) |
04:42.44 | IQ | <PROTECTED> |
04:42.55 | CoolAcid | no , / |
04:42.56 | dogz- | Shido6: sorry didnt understand what u said |
04:43.14 | CoolAcid | exten => 999,1,Dial(Console/dsp) |
04:43.33 | IQ | CoolAcid: thanks :) |
04:43.59 | *** join/#asterisk goobster (goobster@c-67-168-105-166.client.comcast.net) |
04:45.29 | *** join/#asterisk dersteer (~travis@24.231.151.119.gha.mi.chartermi.net) |
04:45.40 | dogz- | Shido6: If you were asking where accepting incoming calls, i mean from my POTS connection (channel 4) |
04:45.56 | *** join/#asterisk coppice (~chatzilla@209.203.17.210.dyn.pacific.net.hk) |
04:46.27 | CoolAcid | Damn.. ipv6 is working.. really slow to pastebin.ca :) |
04:46.49 | dogz- | =] |
04:47.47 | CoolAcid | I have no work with FXO's but shouldn't signalling=fxs_ks read something about FXO not fxs? |
04:48.10 | CoolAcid | erm.. no support with zaptels at all :) |
04:48.47 | dogz- | I thought with the analog connections you had to inform asterisk what function it was perviding |
04:48.57 | dogz- | providing* |
04:48.59 | bkw_ | moving on |
04:49.04 | bkw_ | you signal FXO with FXS |
04:49.07 | bkw_ | and FXS with FXO |
04:49.08 | bkw_ | NEXT!!! |
04:49.28 | dogz- | woohoo did something right :) |
04:49.30 | dogz- | i dink |
04:49.35 | bkw_ | dink? |
04:49.36 | CoolAcid | No one was answering.. Sorry to help. |
04:49.40 | CoolAcid | think. |
04:50.08 | dogz- | dink = think, sorry drinking atm |
04:50.36 | bkw_ | YAY |
04:50.39 | bkw_ | i'm crankin my ipod up |
04:50.43 | bkw_ | trying to brain storm |
04:50.48 | dogz- | bkw_, u mind taking a look at my pastebin? |
04:50.55 | dogz- | http://pastebin.ca/8262 |
04:50.56 | bkw_ | I can try |
04:51.05 | dogz- | thank you |
04:51.11 | bkw_ | LET ME GUESS |
04:51.16 | dogz- | im having issues accepting calls |
04:51.17 | bkw_ | you're using 2.6.11? or .10 |
04:51.19 | dogz- | over the fxo |
04:51.20 | bkw_ | something higher than 2.6.9 |
04:51.24 | bkw_ | kernel |
04:51.30 | dogz- | freebsd actually |
04:51.35 | bkw_ | move along.... |
04:51.45 | cypromis | lol |
04:51.52 | bkw_ | can't help or even begin to help on a platform thats not offically supported... |
04:52.05 | bkw_ | I like hardware that somewhat works |
04:52.17 | bkw_ | but running in on FREEBSD = hardware that might never work right |
04:52.18 | cypromis | you o ? |
04:52.22 | cypromis | where do you find it ? |
04:52.23 | dogz- | =p aight, thanks anyways |
04:52.31 | bkw_ | I see you lookin at me like I got what ya need |
04:52.51 | bkw_ | I wish I could help |
04:52.59 | bkw_ | but the freebsd stuff differs a bit from linux |
04:53.00 | Katty | bkw_: you're just in time to say g'night! |
04:53.11 | bkw_ | Katty, I LOVE YOU!!!!!!!! |
04:53.15 | Katty | oh |
04:53.19 | Katty | uh |
04:53.21 | Katty | k |
04:53.24 | bkw_ | you can be my sexless lover |
04:53.32 | dogz- | no problem :) im just waiting for FC3 isos to finish downloading |
04:53.33 | Katty | that sounds fun |
04:53.36 | Katty | does it include muffins? |
04:53.40 | bkw_ | yes |
04:53.44 | Katty | excellent, brt |
04:53.46 | bkw_ | muffins are good |
04:53.58 | bkw_ | I have about 10 sexless lovers.. most of them are men |
04:54.02 | bkw_ | straight men at that |
04:54.03 | IQ | CoolAcid: can't dial Console from SIP extension :( ... Console to SIP works |
04:54.06 | Katty | :< |
04:54.13 | bkw_ | you're the first female |
04:54.16 | bkw_ | hehe |
04:54.20 | Katty | you've obviously insaned |
04:54.29 | Katty | THIS CALLS FOR MUFFINS |
04:54.30 | CoolAcid | what happeneds in cli? |
04:54.31 | *** join/#asterisk NormAst (~NormAst@toronto-HSE-ppp3972900.sympatico.ca) |
04:54.37 | bkw_ | muffins mmmmm.. yummy |
04:54.39 | bkw_ | oh file dear |
04:54.40 | bkw_ | wake up |
04:54.43 | bkw_ | someone is talking muffins |
04:54.48 | Katty | i mean |
04:54.49 | lilneon | hey has anyone got SMS working with asterisk? |
04:54.50 | Katty | bedtime |
04:54.51 | Katty | yes |
04:54.54 | bkw_ | hehe |
04:55.00 | Katty | stop distracting me! |
04:55.06 | Katty | charlie is waiting! |
04:55.08 | bkw_ | lilneon, the SMS in asterisk is euro only |
04:55.16 | lilneon | :( bummer |
04:55.17 | bkw_ | if you're in the US you're SOL |
04:55.18 | Katty | and heaven forbid i keep the kitty waiting |
04:55.27 | bkw_ | see in euro they send callerid before the first ring |
04:55.34 | bkw_ | so you can get away with sending message to phones |
04:56.12 | lilneon | bkw_:ouch.. so the rest of us gotta just sit on our thumbs and lok for other sms solutions? |
04:56.17 | JT19 | newbie here with a question -- I'm looking to replace an old Nortel system in a small office that is used to having 4 line appearances, and everyone placing / picking up calls from on hold... Is there a way to implement this with asterisk and sip? I've got asterisk doing a TON of other cool, stuff, but not sure how to do this.... |
04:56.17 | IQ | CoolAcid: Mar 25 16:53:02 WARNING[6820]: channel.c:2033 ast_request: No channel type registered for 'System' |
04:56.34 | bkw_ | lilneon, their isn't a "true" sms solution |
04:56.49 | bkw_ | IQ don't dial(System/ |
04:56.49 | bkw_ | ninny |
04:56.55 | bkw_ | system is an app |
04:56.57 | bkw_ | not a channel driver |
04:57.01 | bkw_ | :) |
04:57.06 | cypromis | hey babe dial my system |
04:57.10 | lilneon | bkw_:so.. there are only ppl talking to providers smsc's? and charging for it? |
04:57.21 | bkw_ | smsc's are too fucking expensive |
04:57.28 | bkw_ | 5 cents per message my FUCKING ASS |
04:57.36 | lilneon | bkw_:yes they are.. |
04:57.38 | bkw_ | some guy posted an asterisk solution |
04:57.38 | cypromis | depends how many smses you send |
04:57.43 | bkw_ | but its not really an asterisk solution |
04:57.54 | bkw_ | cypromis, ya but I want a fucking connection to the SMSC |
04:57.57 | IQ | bkw_: so I can not dial Console from an extension ? |
04:57.57 | bkw_ | how can I do that? |
04:58.01 | bkw_ | without paying out the ass |
04:58.02 | lilneon | bkw_:ok.. worth a look.. where??? |
04:58.05 | coppice | bkw_: free SMSCs are not expensive, but connecting them to others can be |
04:58.07 | CoolAcid | yes.. as I posted before. |
04:58.11 | cypromis | bkw: run your own smsc |
04:58.19 | bkw_ | coppice, where can I get connected to an smsc? |
04:58.24 | CoolAcid | IQ: exten => 999,1,Dial(Console/dsp) |
04:58.26 | bkw_ | got any links? |
04:58.26 | Newbie___ | anyone try out 2VA VoIP Asterisk yet ? |
04:58.38 | IQ | CoolAcid: No need for 999,2,Hangup ? |
04:58.45 | coppice | bkw_: you must pay and pay and pay.... |
04:58.51 | bkw_ | haha |
04:58.55 | CoolAcid | IQ: You would probably want that. |
04:59.21 | bkw_ | http://www.bayhamsystems.com/asterisk.html |
04:59.27 | coppice | bkw_: I have connected to SMSCs for high volume, but it only works out economic for pretty high volume |
05:00.18 | lilneon | bkw_: yeah i checked that one already..still damn expensive.. ROI not worth it.. not doing too much bulk..or not consistent with bulk number of msg's |
05:00.36 | bkw_ | 0.05356 per message |
05:00.46 | bkw_ | wonder if you can receive messages |
05:00.56 | bkw_ | I wanna get hooked up to receive messages from the smsc |
05:01.00 | coppice | yes, you can |
05:01.04 | IQ | CoolAcid: exten => 999,1,Dial(System/dsp) |
05:01.04 | IQ | exten => 999,2,Hangup |
05:01.14 | CoolAcid | sure |
05:01.16 | bkw_ | System/ is not valid |
05:01.32 | CoolAcid | Arg.. missed that. |
05:01.45 | IQ | bkw_: so its not possible to dial Console from an extension ? |
05:01.45 | bkw_ | hehe |
05:01.49 | anachron | in asterisk on freebsd where are sounds stored? |
05:01.52 | bkw_ | ya if you do Console/ |
05:01.55 | bkw_ | and not System/ |
05:02.01 | bkw_ | look at that for a moment |
05:02.02 | IQ | bkw_: Okay, thanks |
05:02.03 | bkw_ | let it sink in |
05:02.06 | bkw_ | ;) |
05:02.25 | bkw_ | coppice, where is a cheap way to get messages from the sms network? |
05:02.42 | bkw_ | anyone have the at command set to send SMS's with a motarola? |
05:02.58 | bkw_ | hehe |
05:03.12 | coppice | bkw_: I can give you software for free, but they will charge to send you the messages |
05:03.23 | bkw_ | coppice, thats fine with me |
05:03.25 | bkw_ | I don't mind paying |
05:03.36 | bkw_ | if its not over 4-5 cents a message |
05:03.47 | coppice | bkw_: all phones use the same commands - its a part of the ETSI specs. |
05:04.03 | bkw_ | ok I can find it then I suspect |
05:04.06 | IQ | bkw_ , CoolAcid its working now with 999,1,Dial(Console/dsp) - thaks |
05:04.20 | *** join/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net) |
05:04.25 | bkw_ | newl, t-mobile? |
05:04.44 | Qwell | Do people in the US actually use SMS? |
05:04.50 | bkw_ | I do like crazy |
05:04.54 | newmember | does the FXO module on the TDM400P handle distinctive ring? |
05:04.55 | newl | bkw_: Nope. I work for the largest telco in the southern hemisphere. :) |
05:04.57 | Qwell | You're special though |
05:05.14 | coppice | bkw_: we used some unlimited SMS accounts, but found after about 20,000 per week they were not really free :-) |
05:05.50 | *** join/#asterisk dersteer (~travis@24.231.151.119.gha.mi.chartermi.net) |
05:06.07 | lilneon | hey guys, know anywhere i can get a tdm400p card for less than 305US? FXS only.. |
05:06.14 | IQ | I've SprintPCS unlimited SMS... but they dont go outside US/Canada :( |
05:06.22 | lilneon | or am i better of with a channel bank and a X100p? |
05:06.39 | PatrickDK | channelbankand x100p? |
05:06.45 | jayeola | on my phone network, called "3", i've just discovered that sms to nigeria are not free! |
05:06.49 | mikegrb | jiop' |
05:06.52 | mikegrb | iho |
05:06.56 | mikegrb | I'm nobdrunk |
05:07.02 | PatrickDK | your better off with a channelbank but you need a t1 or e1 for that |
05:07.11 | bkw_ | coppice, hehe |
05:07.17 | mikegrb | channel bnakl is bestg |
05:07.18 | bkw_ | coppice, so how can I receive these SMS's? |
05:07.42 | lilneon | PatrickDK: what if i just wanna connect some analog phones to my asterisk box.. and perhaps 4 or so POTs from the local telco |
05:08.01 | mikegrb | lilneon: just get channnnel bnak |
05:08.12 | coppice | bkw_ if you are american surely you believe SMS is just a figment of european's imaginations :-) |
05:08.13 | mikegrb | uoy owant expand n fthei futrer |
05:08.34 | bkw_ | coppice, na |
05:08.39 | lilneon | mikegrb:.. english please?..? |
05:08.43 | mikegrb | bkw_: I want lvoe you |
05:08.45 | lilneon | yeah i want to expand in the future |
05:08.49 | PatrickDK | if you are going at more than probably 6 lines,I would go channel bank |
05:08.50 | newl | lysdexic. :) |
05:08.53 | mikegrb | lilneon: chanlke bank |
05:08.57 | PatrickDK | then you just plug fxo or fxs into the channel bank |
05:08.58 | mikegrb | mmm beart |
05:09.31 | lilneon | but wont i still need a digium card or something to connect that channel bank to my asterisk box/boxes? |
05:09.44 | PatrickDK | ya, t1 or e1, card |
05:09.56 | newl | Or a BRI card. |
05:09.59 | PatrickDK | unless you use a different device to connect the channelbank to anetwork |
05:10.09 | PatrickDK | hmm, channelbank over bri, how evil |
05:10.15 | newl | 8) |
05:10.22 | lilneon | PatrickDK:that would depend ojn the channel bank i get.. |
05:10.28 | CoolAcid | JT19: just shoot the question. |
05:10.37 | PatrickDK | that, or really what country your in |
05:10.39 | JT19 | I'm looking to replace an old Nortel system in a small office that is used to having 4 line appearances, and everyone placing / picking up calls from on hold... Is there a way to implement this with asterisk and sip? I've got asterisk doing a TON of other cool, stuff, but not sure how to do this.... |
05:10.49 | PatrickDK | ulaw works better in usa than alaw does |
05:11.00 | PatrickDK | cause of the different properties of them |
05:11.02 | lilneon | PatrickDK: in caribbean.. |
05:11.13 | PatrickDK | you use ulaw or alaw down there? |
05:11.13 | CoolAcid | JT19 -- look at call parking |
05:12.05 | JT19 | Is there anyway to show the status of the lines on the actual phones? |
05:12.10 | lilneon | PatrickDK: honestly i don't know.. |
05:12.24 | PatrickDK | look it up online |
05:12.27 | JT19 | I would like the functionality and features of asterisk, but I need it to still be like a "key" system... |
05:12.30 | PatrickDK | ulaw=t1 alaw=e1 |
05:12.56 | lilneon | PatrickDK: well t1.. :D.. haven't heard anything about e1's down here.. |
05:13.15 | lilneon | most companies get T1's, t2's or t3's.. |
05:13.18 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
05:13.57 | *** join/#asterisk Sedorox (~Sed@Neptune-W.client.wlgrv.pa.sed6.net) |
05:14.23 | *** join/#asterisk riksta (~rick@81-178-199-213.dsl.pipex.com) |
05:14.23 | CoolAcid | JT19 - that i'm not sure. |
05:15.06 | JT19 | ok, thank you for the info |
05:15.42 | IQ | How can I send Caller's Info when dialing from Console? |
05:16.13 | JT19 | I was hoping i would find someone who was trying to do what I am. I've found a few phones that say they have "line appearances", but I wasn't sure if that is exactly what I needed. |
05:16.31 | CoolAcid | IQ: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20SetCallerID |
05:16.58 | IQ | CoolAcid: this will work when dialing from console too ? |
05:17.10 | CoolAcid | IQ: Should.. it sets the CallerID of the call.. |
05:18.31 | lilneon | ok.. so back to my question.. where can i get a T1card at a good price? |
05:18.53 | *** join/#asterisk blueskiesokie (~blueskies@65.242.87.151) |
05:18.59 | IQ | CoolAcid: I've used it for extensions - but never for console, not sure how it will work with console... hmmm |
05:19.00 | newl | lilo: Checked ebay? |
05:19.15 | newl | meh..nick completion |
05:19.18 | CoolAcid | IQ: So try? if it doesn't work, try something else. ;) |
05:19.31 | IQ | CoolAcid: :) yeah |
05:21.13 | bkw_ | OMG someone send me about 40k |
05:21.40 | brc_ | no |
05:22.52 | tainted- | bkw_ for what? |
05:23.46 | brc_ | JT19, what you want are snom phones |
05:23.59 | brc_ | then you will need to look into the 'hint priority' |
05:24.12 | brc_ | but honestly, if you want a key system you probably won't be happy with asterisk |
05:24.44 | coppice | if you want a key system, you probably won't be happy with a phone system :-) |
05:24.47 | brc_ | unless people are willing to change and learn how to use a pbx |
05:24.48 | IQ | CoolAcid: confused... dont know exactly where to call this method for Console :P |
05:25.01 | CoolAcid | in your dialplan |
05:25.06 | brc_ | transfer calls instead of telling somebody to pickup line 2 |
05:25.08 | brc_ | etc |
05:25.09 | CoolAcid | before the dial() |
05:25.14 | Moc | JT19, Polycom IP phone does Line apperance too |
05:25.23 | brc_ | does it actually work? |
05:25.24 | lilneon | hey guys, would u recommend the X100P card? since digum has disconinued them.. |
05:25.44 | dmccollum | I just installed one in my system and it works fine. |
05:25.46 | brc_ | they are okay for playing around with |
05:26.07 | Moc | my big issue with analog card are Sidetone !!! |
05:26.26 | IQ | CoolAcid: "NO CALLER ID" - thats what my phone says :P |
05:27.09 | lilneon | brc_: how bout planning for actual deployment?.. |
05:27.14 | CoolAcid | IQ: Did you try setting the other CallerID functions? |
05:27.22 | IQ | CoolAcid: actually, I'm confused about where in dialplan this will go. |
05:27.22 | brc_ | lilneon, I wouldn't use em |
05:27.33 | CoolAcid | Before the dial() |
05:27.37 | IQ | CoolAcid: Yeah, for my other extensions and it work fine |
05:27.40 | brc_ | but then again I don't have any deployments with less then a T1 |
05:30.43 | IQ | CoolAcid: exten => 999,1,SetCallerID(1234567)) |
05:30.43 | IQ | exten => 999,2,Dial(Console/dsp) |
05:30.43 | IQ | exten => 999,3,Hangup |
05:30.46 | lilneon | brc_: ok here's what i need.. a channel bank(6-8fxa / 4fxo) and T1 card for asterisk box.. ebay seems kinda hot.. got anywhere with good prices other than ebay? |
05:31.03 | brc_ | for what |
05:31.04 | brc_ | the cb? |
05:31.32 | CoolAcid | IQ: how about setcidname |
05:31.57 | IQ | CoolAcid: tried that too... also setcidnum |
05:32.10 | lilneon | brc_:um yeah.. and well i plan to move to t1 straight after a while |
05:32.29 | brc_ | dunno...ebay... maybe atacomm.com |
05:33.01 | brc_ | and if I may make a suggestion...buy a 4 port t1 card |
05:33.13 | brc_ | you will wish you had eventually |
05:34.03 | lilneon | brc_: yeah but those kinda HOT!!!... |
05:34.14 | brc_ | hot? |
05:34.15 | IQ | tried all three methods, but still can't send CID from console :( |
05:34.21 | lilneon | brc_:expensive |
05:34.27 | brc_ | BWHHAHA HAHAH AHAHHAHAHAHAH |
05:34.35 | brc_ | no. |
05:35.03 | lilneon | brc_: yeah on ebay those are like not less than $1200 |
05:35.20 | brc_ | it's a decent price for a 4 port t1 card...go compare with a 4 port t1 card for any other system |
05:35.29 | brc_ | and buy from digium.com |
05:37.19 | lilneon | sigh.. now all i need is a good broadband provider and i will be up and running in a month.. any ideas guys? |
05:39.04 | anachron | does someone have a few minutes to help me get zaptel working on my FreeBSD box? |
05:40.28 | *** join/#asterisk Moc_ (~Moc@modemcable012.47-80-70.mc.videotron.ca) |
05:40.36 | *** join/#asterisk jeffik (~jeffik@69.158.33.127) |
05:46.37 | *** join/#asterisk cactus1 (~fasdfsdfa@179.118.33.65.cfl.res.rr.com) |
05:46.53 | cactus1 | anyone want to give me a quick hand? |
05:47.53 | IQ | in sip.conf, if we set fromuser=xyz, it'll be sent to the called device as CID ? |
05:48.09 | newl | Will a SIP client calling out to PSTN stay within its context until the call terminates? i.e. /* context stuff */ Dial(foo) /* do some more stuff here before ending context */ |
05:48.10 | cactus1 | im not sure |
05:48.43 | *** join/#asterisk riksta (~rick@81-178-199-213.dsl.pipex.com) |
05:49.21 | cactus1 | i installed asterisk from an iso i burned, and when i try to log into the web portal it asks me for a password, i tried root and all that the basic stuff and it wont log me in |
05:49.34 | cactus1 | im not sure what to do |
05:50.56 | newl | erm..Asterisk doesn't come with a web portal. Are you using something like Asterisk@home? If so, checking the docs, faq, etc on their site would be my first stop. |
05:51.25 | cactus1 | yea im using @ home |
05:51.39 | cactus1 | and i printed the guidebook and it says nothing about the default web portal password |
05:51.51 | Cherebrum | http://www.wingtunes.com/public/songs.aspx |
05:54.37 | cactus1 | oh well |
05:56.28 | cactus1 | i thought this channel would be of some type of help |
05:56.36 | lilneon | catcus1: (To access AMP use user:maint, pass:password) |
05:56.50 | lilneon | try that one b4 u leave |
05:56.58 | cactus1 | alright |
05:57.18 | lilneon | or root:root and password:password |
05:57.19 | cactus1 | how did you aquire that knowledge? |
05:57.26 | cactus1 | the maint worked |
05:57.33 | cactus1 | i tried root already |
05:57.34 | lilneon | on asterisk wiki |
05:57.43 | lilneon | hold on i bookmarked the page |
05:57.54 | lilneon | http://www.voip-info.org/tiki-index.php?page=Asterisk+at++Home |
05:58.06 | lilneon | seems the default passwords have been hacked or something like that |
05:59.21 | cactus1 | hmm |
06:00.10 | cactus1 | lilneon can i use a voicemodem as a device on my asterisk box? |
06:01.13 | cactus1 | or anyone that wants to answer |
06:01.14 | lilneon | cactus1:not too sure.. i think so have never tried it though |
06:02.07 | cactus1 | what type of device would you recommend getting so i can hook this thing up to a pots line |
06:03.48 | lilneon | um.. a X100p from digium |
06:03.58 | lilneon | or some card like that.. |
06:04.18 | cactus1 | hmm |
06:04.37 | cactus1 | right now im going to try and figure out if i can just get away without buying anything ;) |
06:04.44 | lilneon | yes u can |
06:04.57 | *** join/#asterisk Daishi (~daishi@pool-162-83-251-165.ny5030.east.verizon.net) |
06:04.59 | cactus1 | how so |
06:05.03 | *** part/#asterisk Daishi (~daishi@pool-162-83-251-165.ny5030.east.verizon.net) |
06:05.04 | lilneon | u can use voip providers |
06:05.07 | lilneon | softphones etc |
06:05.09 | cactus1 | packet8 sucks and isnt sip compatible |
06:05.30 | lilneon | asterisk uses many protocols u know.. |
06:05.44 | lilneon | iax,h323 etc |
06:06.11 | *** join/#asterisk riksta (~rick@81-178-199-213.dsl.pipex.com) |
06:06.21 | cactus1 | this stuff is complicated |
06:07.11 | Shido6 | no it isnt |
06:07.19 | Shido6 | what do you want to do ultimately |
06:07.48 | cactus1 | i want to be able to call my landline from my cell |
06:07.54 | cactus1 | and call people on FWD |
06:07.58 | cactus1 | or call out on my packet8 line |
06:08.01 | cactus1 | and have voicemail |
06:08.11 | cactus1 | im using asterik @ home |
06:08.15 | cactus1 | asterisk* |
06:08.46 | cactus1 | i want my asterisk computer to allow me to call FWD people |
06:08.57 | cactus1 | but from any phone, like i call in |
06:09.06 | cactus1 | ya know? |
06:09.24 | lilneon | hmm sounds cool enough.. |
06:09.47 | lilneon | should be able to do that once u get a card.. |
06:10.19 | lilneon | so.. landline->card (on asterisk box)-->internet(FWD,packet8 etc).. |
06:10.33 | cactus1 | they are cheep on ebay |
06:10.39 | cactus1 | yea basically |
06:10.41 | lilneon | relatively |
06:11.40 | cactus1 | its getting bought tommorow |
06:11.49 | cactus1 | it looks just like a modem though |
06:13.14 | cactus1 | hey lilneon technically i could plug my packet8 line into the back of that x100p and pretend like its a POTS line |
06:17.20 | bparker | anyone in here have any experience with Netlogic's IAX service |
06:20.30 | lilneon | cactus1:was away.. um.. technically.. i guess.. but ask the guys on here tomorrow.. wen most of em are awake.. i have never done it |
06:20.52 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
06:21.00 | lilneon | i connect my local telco pstn to my asterisk box and got voip provider to call long distance |
06:22.42 | cactus1 | a voip provider? |
06:22.49 | cactus1 | why not just use the voip phone itself? |
06:23.11 | *** join/#asterisk pulu (~chatzilla@64.200.224.158) |
06:24.31 | lilneon | cactus1: i need the voip provider to terminate my calls oversees.. |
06:25.03 | cactus1 | well dont you have like an adapter that plugs into your router |
06:25.07 | cactus1 | and then plugs into a phone |
06:25.34 | cactus1 | vonage, voicepulse, and packet8 all give you that stuff |
06:26.22 | lilneon | cactus1:yeah basically.. u use them to terminate your calls over seas.. i have slightly more needs than just one analog phone connected to my asterisk box |
06:27.47 | cactus1 | alright |
06:27.50 | cactus1 | well im going to go to bed now |
06:27.55 | cactus1 | night |
06:29.34 | IQ | Hi, How can I override user's entered "User Name" ? Like, I dont want SIP clients to be able to change their names, is it possible? |
06:32.19 | *** join/#asterisk sudhir492 (~sudhir@wbar1.wdc2-4-8-141-004.wdc2.dsl-verizon.net) |
06:32.23 | sudhir492 | hi there |
06:33.20 | IQ | hi |
06:36.23 | *** join/#asterisk rvhi (~rv@66.175.65.89) |
06:37.02 | rvhi | in dial by directory, can i have to try both first and last name? |
06:37.09 | rvhi | currently seems to be either or |
06:38.33 | *** part/#asterisk lilneon (~tj_r3@cuscon12874.tstt.net.tt) |
06:41.36 | *** join/#asterisk NewSole (david@i216-58-44-245.avalonworks.net) |
06:44.38 | IQ | How to override SIP client's "User Name"? I dont want them to change the name I assign - help ? |
06:45.23 | NewSole | in sip profile |
06:45.26 | NewSole | on server |
06:45.35 | NewSole | use |
06:45.46 | IQ | on server side |
06:45.55 | NewSole | callerid = Some User <6665551212> |
06:46.11 | IQ | in sip.conf ? |
06:46.28 | NewSole | yup |
06:47.39 | IQ | NewSole: It works... appreciate your help! |
06:47.46 | NewSole | np |
06:48.10 | IQ | NewSole: would you have a link to all available sip.conf commands? Thanks again for the help :) |
06:48.54 | NewSole | http://www.voip-info.org/wiki-Asterisk+config+sip.conf |
06:49.12 | IQ | Thanks |
06:50.09 | NewSole | np... wish I was as lucky to get help with what I need... |
06:50.43 | IQ | I wish I can help... ask for my life :) |
06:52.00 | NewSole | well my problem is not with asterisk its self.... I am making a Module that does call routing and accounting for prepaid and post paid systems |
06:52.02 | Shido6 | heh |
06:52.10 | Shido6 | yeah thats consulting |
06:52.15 | Shido6 | and ppl will charge you |
06:52.33 | Qwell | cha-ching :p |
06:53.04 | IQ | NewSole: C++ ? |
06:53.35 | NewSole | Shido6... even that would not help... I am sure the $$ is there the backers like what I am doing and want it done fast... |
06:53.50 | NewSole | its just working out the bugs |
06:54.54 | *** join/#asterisk clh (~clh@adsl-69-107-18-183.dsl.pltn13.pacbell.net) |
06:56.57 | NewSole | Shido6... what are you up to tonite |
06:57.03 | IQ | NewSole: what kind of routing? |
06:57.13 | Shido6 | chillin wiff the fam |
06:57.17 | clh | as far as the topic is concerned, shouldn't the person that broke it fix it? |
06:57.35 | Shido6 | then returnin phone calls in the morning |
06:57.42 | Shido6 | knock out a few sessions |
06:58.04 | Shido6 | and write up a marketing plan for hosted services and new products |
06:58.08 | NewSole | well.. I created a database with 4 different providers.. and the module finds the cheapest provider and routes the call |
06:58.12 | Shido6 | then execute in the new week |
06:58.47 | *** join/#asterisk laternight (laternight@cpe-68-201-251-41.houston.res.rr.com) |
06:58.55 | clh | laternight : hi there |
06:58.59 | NewSole | I spent the last 6 months doing that Shido6... finding investor |
06:59.01 | laternight | hi there |
06:59.02 | IQ | NewSole: can't we just create a dial-plan to do all this ? |
06:59.22 | *** join/#asterisk Martz (Martz_test@62.3.201.11) |
07:00.33 | *** join/#asterisk Los415 (~los415@c-24-126-63-233.hsd1.ca.comcast.net) |
07:00.33 | *** join/#asterisk Los415 (~los415@c-24-126-63-233.hsd1.ca.comcast.net) |
07:00.34 | NewSole | ya but problem is it has to manage the 4 providers and the rates change on a daily basis |
07:01.10 | NewSole | the database is updatd evey nite at 4am |
07:01.27 | IQ | NewSole: can't we update the dial-plan at 4:15am - automatically :P |
07:02.17 | NewSole | ya... you could... but in my case I would have to update 5 systems every nite... |
07:02.36 | IQ | NewSole: 5 systems with same dial-plan, right ? |
07:02.37 | Los415 | write a script |
07:02.51 | Qwell | hello, IAX trunking? |
07:02.58 | Qwell | Thats what its for, right? |
07:03.08 | sudhir492 | anyone of you using h323? |
07:03.21 | sudhir492 | what is preferred: h323 or oh323? |
07:03.30 | NewSole | no not same scripts... 3 servers have PRI's and 2 dont |
07:04.28 | IQ | so u need 2 scripts - slightly different |
07:04.30 | sudhir492 | I have been using oh323 since Apr last year and it has been fairly stable. But do not know the status now - i.e. which one is more robust on newer version of Asterisk |
07:07.00 | NewSole | I did not ask for help.... |
07:07.34 | Qwell | Then I have no regrets putting you on ignore. |
07:09.19 | NewSole | o well some peoples kids |
07:10.03 | PTG123 | hey qwell found a fix for the sip problem |
07:10.07 | Qwell | oh yeah? |
07:10.46 | PTG123 | yah |
07:10.55 | PTG123 | you would think i woul dhave been smart enough to save the url though :) |
07:10.56 | PTG123 | one second |
07:10.59 | Qwell | heh |
07:12.14 | PTG123 | http://loans.way2fast.com/patch.txt |
07:12.18 | PTG123 | apply that patch |
07:12.23 | Qwell | I find it funny when you dumb down an explanation for somebody... |
07:12.34 | Qwell | and they turn around and start using technical terms about the subject matter |
07:12.50 | Qwell | I called a coworker today, to ask what my CID was showing up as...dumbed it down a bit |
07:13.00 | Qwell | and he turns around and explains that his internal DID was showing up |
07:13.11 | PTG123 | heh |
07:13.20 | PTG123 | hey what happened to our calling plan today? :) |
07:13.33 | Qwell | family has been here all day |
07:13.42 | Qwell | Sunday? |
07:14.10 | PTG123 | easter? :) |
07:14.15 | Qwell | damn |
07:14.22 | Qwell | Monday it is then |
07:14.25 | PTG123 | i am sure that would work :) not :) |
07:14.41 | Qwell | I don't even know when holidays are coming up... |
07:14.47 | Qwell | we get like...6 |
07:14.50 | PTG123 | you have kids now |
07:14.55 | PTG123 | you gotta know that stuff |
07:14.56 | PTG123 | :) |
07:15.04 | Qwell | yeah, so in about...5 years, when he starts going to school... :p |
07:15.24 | PTG123 | no man the easter bunny is coming tommorow night |
07:15.34 | Qwell | he comes at night? |
07:15.39 | PTG123 | yes |
07:15.44 | Qwell | like santa? |
07:15.49 | PTG123 | and leaves easter baskets and eggs hidden for your son |
07:15.51 | Qwell | That explains the gifts then... |
07:15.56 | PTG123 | you don't know that? :) |
07:16.20 | Sedorox | anyone use lingo? |
07:16.26 | Qwell | have you ever heard of "easter gifts"? |
07:16.30 | PTG123 | yes |
07:16.32 | Qwell | coworker told me about those yesterday |
07:16.35 | PTG123 | heh |
07:16.41 | PTG123 | were your parents athesists? |
07:16.43 | Qwell | I was like, "wtf is an easter gift?" |
07:16.49 | PTG123 | non holiday loving ones at that? |
07:16.54 | Qwell | the easter bunny was the devil :P |
07:16.55 | Sedorox | PTG123: do you have any tips on setting it up? or a howto? |
07:17.04 | Qwell | he killed jesus...or something |
07:17.09 | PTG123 | Sedorox, why lingo? |
07:17.10 | Qwell | I never really paid much attention to my parents |
07:17.11 | Sedorox | or you answering Qwell |
07:17.20 | Sedorox | I'm helping someone set it up... |
07:17.30 | PTG123 | Sedorox, will it even work with asterisk? |
07:17.39 | PTG123 | Sedorox, i would use an asterisk friendly provider |
07:17.41 | Sedorox | dunno |
07:17.48 | Sedorox | its a little late :-p |
07:18.04 | PTG123 | cancel :) |
07:18.14 | PTG123 | Qwell, its your job to spoil that kid |
07:18.17 | Sedorox | aha |
07:18.57 | Qwell | brb, gonna patch this up |
07:19.38 | *** part/#asterisk clh (~clh@adsl-69-107-18-183.dsl.pltn13.pacbell.net) |
07:19.44 | *** part/#asterisk laternight (laternight@cpe-68-201-251-41.houston.res.rr.com) |
07:23.31 | *** part/#asterisk Cheng29 (~cheng29@d57-87-253.home.cgocable.net) |
07:25.57 | Qwell | PTG123: Wanna go to aim? |
07:26.10 | PTG123 | sure |
07:26.15 | PTG123 | oh shit :) |
07:26.30 | PTG123 | nah use irc |
07:26.39 | PTG123 | aim is open underneath countrstrike on my other pc |
07:26.40 | PTG123 | :) |
07:29.10 | Sedorox | how does teliax work for most pople... good? |
07:29.30 | PTG123 | yah its not too bad, what kind of account are you looking for? |
07:29.47 | Sedorox | residential |
07:30.00 | PTG123 | which state? |
07:30.04 | PTG123 | like the unlimited plan or what? |
07:30.17 | Sedorox | MA. and I would assume so.. this is actually for someone else |
07:30.17 | Sedorox | :-p |
07:30.31 | Sedorox | just seeing if people have had luck with it |
07:32.22 | PTG123 | sure try em out :) |
07:32.56 | Sedorox | kk |
07:34.03 | IQ | If we get "Pay as you go" package from Teliax - they charge 2 cent/incoming-call? (A two cents connection charge applies unless call is from another teliax user) |
07:34.42 | Sedorox | hmm |
07:34.56 | PTG123 | ya |
07:35.58 | IQ | not bad |
07:47.53 | *** join/#asterisk pulu (~chatzilla@64.200.224.158) |
08:11.12 | sudhir492 | Anyone isomg chan_h323 here? |
08:11.22 | sudhir492 | using |
08:31.53 | *** join/#asterisk AlexW (~AlexW@adsl-46-159.swiftdsl.com.au) |
08:40.49 | *** join/#asterisk bonez39 (~aint@drjones.dsl.xmission.com) |
08:46.05 | *** join/#asterisk SexyKen (~sexyken@c-67-161-5-149.client.comcast.net) |
08:46.07 | SexyKen | Hey gulys. |
08:46.09 | SexyKen | *guys. |
08:46.14 | SexyKen | Sorry, a little intoxicated. |
08:46.15 | SexyKen | Anywho. |
08:46.18 | SexyKen | Anyone want a job? |
08:47.15 | bonez39 | I need a job..heheh |
08:48.44 | *** join/#asterisk mbranca (~matteo@host-84-222-14-188.cust-adsl.tiscali.it) |
08:54.32 | *** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net) |
08:55.35 | SexyKen | bonez, what can you do for me? |
08:56.17 | Sedorox | does it pay? |
08:56.31 | SexyKen | Yes. |
08:56.42 | Sedorox | hmm |
08:56.47 | Sedorox | involves? |
08:57.05 | SexyKen | Setting up an entire asterisk system |
08:57.19 | Sedorox | for how many people/extentions and hardware? :-p |
08:58.31 | SexyKen | It's a remote job. |
08:58.34 | SexyKen | Work from home. |
08:58.38 | Sedorox | nice |
08:58.45 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
08:58.46 | Sedorox | any T* hardware involved? lol |
08:58.49 | SexyKen | I need many extensions and a GUI to mange everything. |
08:58.50 | SexyKen | No. |
08:58.52 | SexyKen | OIP only. |
08:58.56 | SexyKen | VIPO. |
08:58.58 | SexyKen | VOIP |
08:59.00 | SexyKen | Damn alcohol. |
08:59.01 | Sedorox | nice |
08:59.07 | Sedorox | gui = amp |
08:59.08 | Sedorox | :-p |
08:59.18 | SexyKen | I need 3 companies to be allowed on asteriskl. |
08:59.29 | Sedorox | hell.. I'd be interested in it |
08:59.37 | Sedorox | I run three asterisk boxes right now |
08:59.51 | SexyKen | I need a single asterisk box running 3 companies. |
09:00.02 | Sedorox | pushing a lot... |
09:00.11 | Sedorox | sounds easy enough tho... |
09:00.29 | SexyKen | For instance, companies a, b, c would be able to have separate extenions ( each could have an extension 200, just going to different phones) |
09:00.38 | Sedorox | yea |
09:00.40 | SexyKen | 7k/mintues a week max. (total) |
09:00.42 | newl | Everything sounds easy until you hear the details. :) |
09:00.50 | Sedorox | newl: true |
09:01.14 | SexyKen | Need user/extension management. |
09:01.28 | SexyKen | Need to be able to view complete cdr shit. |
09:01.29 | SexyKen | etc. |
09:01.30 | Sedorox | I'm interested in it... |
09:01.38 | Sedorox | but just don't know the cdr stuff... yet... |
09:01.46 | Sedorox | and the management stuff... |
09:01.48 | SexyKen | Well it's not a 3 month project. |
09:01.51 | Sedorox | only stuff I know of is amp |
09:01.59 | BoRiS | of course |
09:02.00 | Sedorox | I know.. probably more like 2 weeks |
09:02.00 | Sedorox | :-p |
09:02.01 | SexyKen | It's something I needed done 3 months ago. |
09:02.12 | BoRiS | But again...depends on how much :) |
09:02.19 | SexyKen | How much what? |
09:02.20 | Sedorox | thatgs true too |
09:02.21 | Sedorox | :-p |
09:02.25 | Sedorox | to do this |
09:02.29 | SexyKen | $2k. |
09:02.33 | Sedorox | 0_o |
09:02.35 | BoRiS | keep going |
09:02.44 | Sedorox | lol |
09:03.01 | BoRiS | :) |
09:03.08 | Sedorox | I need money.. so... |
09:03.08 | Sedorox | :-p |
09:03.13 | newl | Do four deals like that a month and you'd be a happy camper. 8) |
09:03.36 | Othello | I'm getting pissed off that the console driver is not working for *... |
09:03.45 | SexyKen | I need it done, yesterday. GUI with Comopany Management, Extension Management (add/edit/delete etc) should be able to add IVR's through the stupid gui thing. Should be able to view entire call details based on queues, agents, etc. |
09:03.46 | Sedorox | yea.. really... |
09:04.02 | *** join/#asterisk jskcr (~kvirc@jskcr.user) |
09:04.10 | Sedorox | :-p |
09:04.28 | SexyKen | AMP doesn't work for multiple copmanies ;-) |
09:04.37 | Sedorox | :-p |
09:04.59 | BoRiS | Amg....Ugh |
09:07.05 | Sedorox | BoRiS: you do the gui.. I'll do the rest.. and we split |
09:07.06 | Sedorox | :-p |
09:08.12 | BoRiS | hehe |
09:08.21 | BoRiS | I prefer the rest "other" then the gui |
09:08.35 | Sedorox | lol |
09:16.05 | Sedorox | allwell... |
09:39.08 | *** join/#asterisk _Omer (~eegg@202.38.63.84) |
09:40.10 | _Omer | My question is not related to Asterisk but I want to know. What things are required to develop FREEWORLDDIALUP.COM (FWD) kind of thing?? |
09:48.36 | *** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net) |
09:55.40 | CleanerX | new linphone is out |
09:55.43 | CleanerX | 1.0.x |
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11:32.35 | cjk | hi, can sip an iax RTP streams bridge? i mean, if i have canreinvite set to yes, will the rtp stream go through * ? |
11:51.46 | *** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc) |
11:51.46 | *** topic/#asterisk is Asterisk: The Open Source PBX || 1.0.7 Released || http://www.theregister.co.uk/2005/03/16/asterisk_open_source_pbx/ || cdr_addon_mysql.c with 1.0.7 == DEADLOCK someone that cares needs to fix it.. because I don't (bkw_) |
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12:03.42 | cjk | mbranca, never |
12:03.58 | cjk | mbranca, forget about portforwdings, i never got one phone working because of portforwarding |
12:04.03 | cjk | thats for sure not the solution |
12:04.17 | cjk | mbranca, there are several nats where it just does not work. |
12:04.24 | cjk | but its working on most nat's |
12:04.39 | *** part/#asterisk loxza (~loxza@118.Red-81-33-60.pooles.rima-tde.net) |
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12:07.48 | *** join/#asterisk DannyF (~wizardone@h163n9c1o848.bredband.skanova.com) |
12:08.48 | mbranca | cjk: letting rtp flowing through asterisk made sip devices working behind any kind of nat (at least in my experience) |
12:08.59 | mbranca | of course there's a bw problem |
12:09.09 | mbranca | but depends on what you need to do |
12:09.18 | mbranca | and without any sort of port forwarding |
12:09.38 | cjk | mbranca, thats true for several nats |
12:09.52 | cjk | but its not necessary for many nats |
12:10.06 | mbranca | till now I never experienced any problems |
12:10.22 | mbranca | and we have a rather large customers base :) |
12:13.03 | Mw3 | hi folks, have you seen any ata which supports vpn ? |
12:29.00 | Mavvie | http://www.voip-info.org/wiki-Asterisk+cmd+SrxEchoCan <- /me needs it for SIP channels :-/ |
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13:05.48 | cjk | maybe someone know here, if i do an sip-2-iax call or iax-2-sip (both set to canreinvite=yes) will the voice traffic pass my * or not? |
13:06.17 | cjk | maybe someone knows here, if i do a sip-2-iax or iax-2-sip phone call (both set to canreinvite=yes) will the voice traffic pass my * or not? |
13:06.18 | Zeeek | it would seem logical |
13:06.27 | Zeeek | that it would since the two devices can't talk |
13:07.04 | ckruetze | cjk it will, since you sip phone doesn't speak iax and your iax phone doesn't speak sip |
13:07.50 | *** join/#asterisk mmlj4 (~looseduk@ip68-14-39-201.no.no.cox.net) |
13:07.59 | cjk | yeah but i thought maybe the voicestream would be the same or something like that |
13:08.12 | Zeeek | the ports used are totally different |
13:10.07 | Zeeek | so why do you wonder? |
13:10.15 | Mavvie | well, reload doesn't work on it. |
13:10.33 | Zeeek | try a restart and you'll have the definitive answer |
13:10.40 | Mavvie | oh! reload only does re-read the configuration files. |
13:10.47 | Mavvie | it doesn't reload the module. |
13:11.39 | Zeeek | makes sense. Thank you for sharing that info with us. |
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13:15.14 | *** join/#asterisk Mother_ (~mother@93.Red-80-32-127.pooles.rima-tde.net) |
13:15.18 | Mother_ | multiple greetings |
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13:18.47 | cjk | does sip-2-iax and iax-2-sip translation use alot of cpu? |
13:20.41 | queuetue | I'm trying to set up iax between two servers, and on the north server, I see "Peer 'south peer' is not dynamic (from x.x.x.x)" ... What does this mean? |
13:21.28 | queuetue | The south server reports "registration refused" ... |
13:21.37 | Mother_ | check the host setting in iax.conf |
13:21.43 | Zeeek | cjk my logic may be faulty but if I needed to worry about this, do it all in IAX. Obviously, we can't dictate what protocols users have, though |
13:21.52 | Mother_ | I have mine set to the remote IP |
13:21.58 | Mother_ | hi Zeeek |
13:22.01 | Zeeek | lo ma |
13:22.12 | queuetue | Now, north to south, the connection seems fine, and the two iax.confs are copies of each other with hosts swapped ... |
13:22.28 | queuetue | Mother_, The settings on which server? |
13:22.49 | queuetue | What does s"not dynamic" mean, in this context? |
13:22.54 | Mother_ | on both, I have a similar setup, two servers each behind their respective NATed DSL lines |
13:23.39 | Mother_ | well I'm not sure, I've never seen that message, however I can tell you that setting host=x.x.x.x instead of dynamic doesn't cause me any problems |
13:23.44 | Mother_ | I have fixed IPs however |
13:24.17 | queuetue | Mother_, As do I - neither of these servers is dynamic. |
13:24.38 | Mother_ | and have you got the IP address in the host parameter in iax.conf? |
13:24.43 | queuetue | and I have host=x.y.y.z on both servers. |
13:24.48 | queuetue | yes. |
13:25.04 | Mother_ | and your ports are mapped OK if you have routers/NAT/etc? |
13:25.24 | queuetue | Mother_, Yes, but I will test again. |
13:25.36 | queuetue | The wierd thing is that one connection (north->south) works ... south->north does not... |
13:25.40 | Mother_ | you can use iax2 debug to see what is happening |
13:25.50 | queuetue | Mother_, How do I do that? |
13:25.51 | Mother_ | in the CLI |
13:25.55 | queuetue | Ah. |
13:25.58 | Mother_ | CLI> iax2 debug |
13:28.28 | Mother_ | gotta go eat some lunch |
13:28.30 | Mother_ | bbiab |
13:33.24 | cjk | if i have a sip and an iax client, both doing ilbc, will * have a lot of work to translate? |
13:39.28 | Damin | It is way to early in the morning.. |
13:41.57 | tzanger | this is fucking terrible |
13:42.05 | mmlj4 | does the cheap intel modem that works as an FXO also do the timing that's needed for IAX links? |
13:42.12 | tzanger | all the legal music download sites want you ot register before you can SEARCH their archives |
13:44.31 | Zeeek | tzanger why is that so terrible? |
13:44.46 | tzanger | why should I give anyone my credit card information in order to see if they have the music I want? |
13:45.02 | Zeeek | register means giving cc number? |
13:45.05 | tzanger | if I want to purchase from them, sure. But to search? fuck that. |
13:45.07 | tzanger | yup |
13:45.12 | Zeeek | scew that |
13:45.16 | Zeeek | screw |
13:45.20 | tzanger | "You can cancel if you don't want to buy anything from us" |
13:45.34 | Zeeek | I never give a number unless I know I'm gonna buy and that the place is cool |
13:45.40 | tzanger | exactly |
13:45.41 | Zeeek | sounds BOGUS to me |
13:45.46 | *** join/#asterisk das1 (~dh@cau138.neoplus.adsl.tpnet.pl) |
13:45.55 | Zeeek | no one would do that with *their own* cc number :) |
13:46.05 | tzanger | mindawn.com looks cool but the selection blows |
13:46.10 | tzanger | ogg/flac formats |
13:46.16 | das1 | does someone have an idea on this: 2 asterisk calling each other, one is immediately hanging up after haveing received ack and first ring. 1.0.7 debian SID the not working one |
13:46.31 | tzanger | and itunes wants me to download software to search |
13:46.37 | tzanger | where's bkw or file when you need them |
13:46.48 | tzanger | I bet both those buggers have itunes accounts, they can search that for me |
13:46.57 | mmlj4 | das1: is either behind a NAT connection? |
13:47.06 | das1 | yes |
13:47.18 | mmlj4 | using SIP, or AIX? |
13:47.27 | das1 | but if you start from scratch it's working few minutes |
13:47.31 | das1 | iax |
13:47.40 | mmlj4 | hmm... no clue, then |
13:48.06 | tzanger | allofmp3.com lets me search without registering, but they don't have the artist either |
13:48.08 | das1 | mmlj4: thanks |
13:48.12 | mmlj4 | SIP + NAT = problems, as we know |
13:48.20 | tzanger | SIP + Internte = problems |
13:48.23 | Zeeek | what are you looking for, Perry Como ? |
13:48.25 | tzanger | er internet |
13:48.33 | tzanger | Zeeek: hahaha no, Kevin Laliberte |
13:48.39 | Zeeek | canadian? |
13:48.42 | tzanger | yes |
13:48.50 | Zeeek | try this: virgin.fr |
13:48.57 | Zeeek | pretty good search engine |
13:49.14 | Zeeek | but wait that's not free |
13:49.21 | tzanger | it says I need Win98 to even get on to the site |
13:49.26 | Zeeek | nah! |
13:49.28 | dant | lol |
13:49.28 | tzanger | at least fromw hat I can tell from my broken French |
13:49.35 | Zeeek | no way! |
13:49.43 | jeffik | try jazzmusique |
13:49.48 | tzanger | yup change konqueror's id and it goes |
13:50.37 | Zeeek | that don't have Kevin Laliberte |
13:50.41 | Zeeek | anyway |
13:50.45 | tzanger | uh virgin.fr's search is completely busted |
13:50.57 | Zeeek | works great for me with Firebird |
13:51.00 | tzanger | I put in "Laliberte" in the search and I get 16 pages of hits, none of which have "Laliberte" in them |
13:51.12 | tzanger | they do have Mel Torme though.. hahaha |
13:51.18 | tzanger | jazzmusique? |
13:51.36 | Zeeek | tzanger ya want free or to pay? |
13:51.48 | tzanger | I don't mind paying for the album at all, I just can't seem to find it online |
13:52.18 | tzanger | jazzmusique has streams, not music to buy |
13:52.26 | jeffik | yes streaming although only 3 formats but is free |
13:52.47 | Zeeek | sony connect, msn music club, tiscali music, jam label, uploud, mp3tunes.com |
13:53.00 | *** part/#asterisk das1 (~dh@cau138.neoplus.adsl.tpnet.pl) |
13:53.07 | Zeeek | you can easily record streams...; on WIndows :) |
13:53.22 | jeffik | ok then subscribe to ripcast/shoutcast it's cheap, 100s o genras of streaming and you can captue it using ripcast |
13:53.28 | tzanger | I can easily do it with LInux and mplayer, but I don't want to record streams, I want to download the album |
13:53.43 | tzanger | jeffik: I want to download an album, not a random stream |
13:53.45 | Zeeek | nobody has heard of it around here! |
13:53.56 | tzanger | Zeeek: he's relatively indie |
13:54.10 | tzanger | http://www.kevinlaliberte.com/ |
13:54.20 | Zeeek | http://newmusic.clearchannel.com/artist/kevinlaliberte/song/14500 |
13:54.51 | Zeeek | haha "You must be a New Music Network member to perform that function." |
13:54.51 | tzanger | heh |
13:54.53 | Zeeek | the net ain't what it once was bioz |
13:55.01 | Zeeek | boyz |
13:55.14 | jeffik | boiz |
13:55.25 | Zeeek | <PROTECTED> |
13:55.41 | jeffik | yes all the free services either went to pay or went away |
13:55.44 | tzanger | Peter Smith Quartet? |
13:55.53 | Zeeek | he's on that |
13:55.55 | tzanger | jeffik: like I said, I'm looking to BUY the album |
13:56.12 | Zeeek | http://www.guitarnoise.com/news/2003/20030914.html |
13:56.30 | jeffik | Zeeek: just trying to help you find some music |
13:56.44 | Zeeek | not me - Is the album elation? |
13:56.55 | tzanger | yes |
13:58.25 | tzanger | I can order the physical CD online but that doesn't get it here for tomorrow :-) |
13:58.47 | Zeeek | you're right, it isn't easy to download that CD |
13:58.59 | Zeeek | looks good, too |
13:59.04 | tzanger | I'm going to go to Kitchener today to see if I can't find it on a shelf but it may prove difficult |
13:59.16 | tzanger | yeah I discovered that I like it quite by accident |
14:01.34 | Zeeek | apparently you can download two songs free here: http://newmusic.clearchannel.com/ |
14:01.39 | Zeeek | membership is free |
14:01.41 | tzanger | yeah |
14:01.48 | Zeeek | looks like a decent site actually |
14:03.31 | tzanger | I hate "enter your email address.... now enter it again" |
14:03.34 | tzanger | I just copy and paste |
14:04.01 | Zeeek | me too - some idiot thought that was a good idea |
14:04.10 | tzanger | well for people who can't type, sure |
14:04.25 | Zeeek | especially since my address on those sites is always something-mykey@sneakemail.com |
14:04.33 | tzanger | hehe that is EXACTLY what I do |
14:04.35 | tzanger | EXACTLY |
14:04.39 | Zeeek | new address for each account |
14:04.41 | tzanger | yup |
14:04.53 | Zeeek | the only trouble is remembering it when trying to log in :) |
14:04.58 | tzanger | :-) |
14:05.12 | Zeeek | sneakemail with auto address keys is the greatest thing since asterisk |
14:06.02 | tzanger | heh |
14:06.10 | Zeeek | "Dear FuckSpam-UpYours@sneakemail.com. It has come to our attention..." |
14:06.31 | tzanger | I have my own domain though |
14:06.39 | Zeeek | "Our client, Mr. Fuckspam has passed away leaving you..." |
14:06.41 | tzanger | graylisting is the best |
14:07.05 | Zeeek | I have my domains too, but I never give that out |
14:07.34 | tzanger | hmm that elation tune was the one I really wanted but I still want the rest of them too |
14:07.35 | Zeeek | not since the domain registrars spammers CD captures |
14:07.53 | Zeeek | So sing in to the network thing, they have elation on download |
14:07.56 | Zeeek | sign |
14:08.00 | tzanger | it was funny I just got finished practising guitar for about an hour when I heard that song and I said "damn... I don't sound anything like that" |
14:08.09 | tzanger | yes I know I already downloaded it |
14:08.12 | tzanger | and the other too |
14:08.20 | Zeeek | cool! |
14:08.27 | Zeeek | you play guitar? |
14:09.33 | Zeeek | check this: http://resmo.com/rare/mp3/logjam.mp3 |
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14:13.26 | tzanger | Zeeek: yeah but I'm nowhere near as good as him |
14:14.18 | Zeeek | as Kevin? |
14:14.41 | tzanger | yeah |
14:14.43 | Zeeek | Kevin looks like he's world class - played with a lot of great people |
14:14.45 | tzanger | I have never tried playing latin guitar either |
14:14.49 | Mother_ | as for the giving out a CC to sites, I have one of these "virtual" CCs that's only a number |
14:14.58 | Mother_ | the credit is exactly zero euros |
14:14.59 | tzanger | Mother_: really? that's a good idea |
14:15.16 | *** join/#asterisk sandnigg0r (~crunkuser@66-55-197-254.gwi.net) |
14:15.30 | tzanger | Zeeek: yeah I am taking this lady I'm seeing down to toronto to see him next month |
14:15.33 | Mother_ | yep, and since I can get as many of these virtual CCs as I want, I keep one for these one-off sites and payments |
14:16.02 | tzanger | I knew this song sounded familliar (then and now) |
14:16.04 | Zeeek | Mother_ that's a little like the email I was talking about. How can we get these? |
14:16.09 | Mother_ | when I have to purchase, I recharge the exact amount, this can be done on ATMs and bank branches |
14:16.24 | tzanger | why's logjam seem familliar |
14:16.40 | Zeeek | you listened to logjam? |
14:16.40 | Mother_ | well my bank is http://www.lacaixa.es the only problem you may have is the recharge |
14:17.23 | Zeeek | we have that bank in France too |
14:17.32 | Zeeek | there's one down the street in fact |
14:17.43 | Mother_ | http://portal1.lacaixa.es/Channel/Ch_Redirect_Tx?dest=1-30-10-00000 |
14:17.57 | Mother_ | that's the actual card, "Cybertargeta" |
14:18.17 | Zeeek | very good idea |
14:18.28 | tzanger | it doesn't bother me |
14:18.41 | tzanger | at least in canada and the us you are only liable for $50 on your CC if hte number's been stolen |
14:18.48 | Mother_ | only problem is the initial recharge has to be done at an office or ATM of this bank |
14:18.48 | tzanger | you don't need to pay extra insurance or anything |
14:18.57 | tzanger | if someone racks up $20000 on your card you pay $50 and that's it |
14:19.14 | Mother_ | that's not bad, here you either buy extra insurance for the card, or it's up to the time when you report it stolen |
14:19.26 | tzanger | *nods* |
14:19.42 | Mother_ | I have a friend who lost some $4000 and was able to get some back only after suing the bank |
14:20.07 | robl^ | tzanger: only for *now* part of the new Bankruptcy re-form the current administration is tryin got pass includes removal of some of that protection.. and prevents you also from removing creditcard debt if you file bankruptcy |
14:20.58 | Zeeek | there must be some advantage to living in Canada now that there's no draft in the US anymore :) |
14:21.07 | Mother_ | lol |
14:22.00 | Zeeek | I suppose one might say that Canada combines the best parts of UK and American culture... which is little the opposite in Oz :) |
14:22.30 | Zeeek | just kidding guys... no offen[cs]e |
14:23.01 | Zeeek | Mother_ you're in Spain, eh? I never noticed |
14:23.10 | Mother_ | yep :) |
14:23.58 | Blackvel | what is the easy way to send "ANSWER" inside AGI? |
14:24.07 | robl^ | Zeeek: true! at least Canada doesn't have Cricket. that game is evil. each game takes an average of 7.23 years to complete.. just wathing guys running back and forth with little sticks shoved in the ground |
14:24.22 | Mother_ | slight doubt: what would be a good tool to batch convert PCM recordings to mp3 that I could stick in cron? |
14:25.08 | tzanger | robl^: yeah i know |
14:25.26 | Mother_ | robl^: agreed, I fell asleep the two times I tried to watch a cricket match |
14:25.34 | tzanger | robl^: the US is heading to hell in a handbasket and awfully damn fast too... it's like the city's on fire and nobody's able to see the flames |
14:26.24 | *** join/#asterisk Chuji (Chuji@pcp09930052pcs.tulipgrove.tn.nash.comcast.net) |
14:26.36 | robl^ | tzanger: I think plenty of people see it, but are put into a state of "terra" and are too scared to do anything about it |
14:26.39 | tzanger | I dunno... so much is on credit these days it can only go on so long it's just like the dot-com bubble |
14:28.01 | *** join/#asterisk Frantic (~ab@24-193-46-85.nyc.rr.com) |
14:28.11 | tzanger | I mean I'm not TOOOOOO bad off, I have a line of credit and a mortgage... that's it. my credit card debt is miniscule (maybe $500). If things were to go sour I'd lose the house and that might be nasty, especially if the real estate market pops, but I don't think it'd be toooo terribly bad. I have *land*, not just a house, and property has intrinsic value. |
14:28.49 | tzanger | I don't have much land, but it is enough to build 6-8 houses on by today's standards, so that's a saving grace |
14:29.51 | newl | Subdivide and conquer! |
14:30.12 | tzanger | hehe I'm 4km from town |
14:30.18 | robl^ | tzanger: well.. what if you loose yer job and a family member gets very sick and has to be in the hospital for an extended stay? a lot of credit debt in the US at this point is due to lack of healthcare insurance for so many people. |
14:30.21 | tzanger | you could do it but yeah |
14:30.39 | Zeeek | plant a server farm! |
14:30.43 | Mother_ | here they stats are that only 35% of people can save up something at the end of the month |
14:30.50 | Mother_ | the rest just manage to get by |
14:31.02 | tzanger | robl^: lose my job, yes that's a problem. I'm trying to get the LoC paid down so I can indeed survive at my current rate of spending for four months. family member sick -- I have insurance for that |
14:31.10 | Mother_ | and the trend is visible when you see TV ads for cars where you can pay them in no less than 9 years!!! |
14:31.20 | tzanger | Mother_: yes that is spooky |
14:31.51 | Mother_ | I mean a car will be worth 1/2 a few months after you buy it....that's just wrong |
14:31.59 | tzanger | yup |
14:32.11 | tzanger | I've never purchased new and likely never will due to the extreme depreciation |
14:32.21 | robl^ | tzanger: if you loose your job, you likely loose healthcare benefits too.. that's the problem. |
14:32.24 | Mother_ | I already know cases of people who have had the car stolen or wrecked and will have to repay it during a few more years, while having to buy another car |
14:33.14 | tzanger | robl^: I have health insurance through the company that is true, but my mortgage is insured and my life insurance covers *my* health problems to a certain degree. i am not saying I'm safe by any means but I do have some coverage |
14:33.46 | Zeeek | fwiw, our coverage goes on for at least a year during unemployment |
14:34.00 | Zeeek | it's state operated, not private |
14:34.01 | robl^ | tzanger: that's good to h ear. unfortuantely you are much better off than a large number of americans |
14:34.21 | tzanger | *nods* I realize that and I do try to work hard to maintain it. :-) |
14:34.27 | tzanger | there are many, many who are worse off than me |
14:34.37 | blitzrage | what is a keytree, and how does it differ from a key family? |
14:34.53 | Zeeek | that's where the keys grow, blitz! |
14:35.31 | Chuji | blitzrage : Does this look like an Asterisk help channel? It's debt/motivation channel... sheesh |
14:35.41 | Zeeek | yeah! |
14:35.49 | blitzrage | oops.. sorry |
14:35.57 | Blackvel | 510 Invalid or unknown command |
14:36.02 | Zeeek | do your own research, blitz! There is a site that has some good stuff.... |
14:36.11 | Chuji | /join #a$terisk |
14:36.12 | Zeeek | some .org thingie |
14:36.16 | Blackvel | why does this happen? I am sending the command "ANSWER" of the AGI channel |
14:37.02 | blitzrage | Zeeek: people who say to go and do their own research say so because they don't know the answer :D |
14:37.08 | blitzrage | Zeeek: it's coming |
14:37.19 | Zeeek | nah, when I say it, I give the answer too - I'm no crtich ! |
14:38.00 | Zeeek | I owuld like to announce a monument day today, a landmark in asterisk history |
14:38.20 | Chuji | heh, the legacy of critch |
14:38.29 | Zeeek | I got the answer to a problem I repeatedly submitted to EDigium support and here for weeks...; on the mailing list! |
14:38.44 | blitzrage | what was it? |
14:38.59 | Zeeek | JEEZE |
14:39.07 | Zeeek | the question or the answer? |
14:39.22 | Zeeek | the question was, why no callerid on ONE recent phone |
14:39.36 | Zeeek | the answer was, change the ring cadece |
14:39.41 | Zeeek | cadence |
14:40.01 | blitzrage | hrmmmm, crazy |
14:40.09 | blitzrage | damn non-north americans |
14:40.11 | blitzrage | ;) |
14:40.13 | *** join/#asterisk Landim (~daniel@200.157.72.19) |
14:40.23 | Zeeek | as to your query blitz, I'd have guessed they were the same thing written by different people. |
14:40.36 | Zeeek | family here, tree there |
14:40.43 | blitzrage | Zeeek: possible |
14:41.06 | Zeeek | unless there are multiple trees of which the / we see is only one? |
14:41.29 | Zeeek | I am not intimate enuf with source to know |
14:42.38 | Landim | Hi All, I need some help with Zap Quad E1 card configuration. When I try to make a call I receive this msg: Unable to create channel of type 'Zap' |
14:42.39 | Landim | <PROTECTED> |
14:43.08 | Zeeek | blitz : snprintf(prefix, sizeof(prefix), "/%s/%s", family, prefix); |
14:43.32 | Zeeek | <PROTECTED> |
14:43.37 | Mother_ | maybe everyone *is* busy and congested |
14:43.41 | Blackvel | I try to send over AGI ANSWER + NEW_LINE. my java application sends ANSWER\r\n instead of ANSWER\n |
14:43.44 | Zeeek | I've been congested recently |
14:43.46 | Blackvel | why is that a problem? |
14:43.50 | Landim | But all are free. |
14:44.02 | Mother_ | me too |
14:44.12 | Landim | I'm setting up the system right now! |
14:44.20 | Mother_ | btw I found the notlame encoder, it works off the command line and is very fast |
14:44.32 | Zeeek | and has a great name |
14:45.02 | Mother_ | so now I can compress all the call recordings to mp3 at night when the system is idle :) |
14:47.56 | Chuji | Mother_ : From what to mp3? |
14:50.04 | *** join/#asterisk fenlander (~irc@82.152.81.57) |
14:50.42 | Mother_ | Chuji: PCM recordings |
14:50.55 | Mother_ | from calls |
14:51.08 | *** join/#asterisk logicalonline (~logicalon@border.logicalonline.com) |
14:51.17 | Blackvel | how can I send this over AGI: EXEC DIAL(ZAP/g1/90,30,tr)? |
14:52.11 | Mother_ | dwmw2: then why don't you setup a cron that compresses and then emails? |
14:52.30 | dwmw2 | Mother_: I meant voicemail. It's emailed automatically by app_voicemail isn't it? |
14:52.35 | dwmw2 | I just need to implement format_ogg |
14:52.59 | dwmw2 | but I'll do that after I have chan_bluetooth working properly |
14:53.26 | Mother_ | hmmkay |
14:53.42 | Mother_ | you can still do without the built-in email notifications and cook your own |
14:53.48 | dwmw2 | true |
14:53.50 | Chuji | dwmw2 : You working on chan_bluetooth? |
14:53.56 | dwmw2 | Chuji: yeah |
14:54.02 | dwmw2 | I have it working with * CVS and full-duplex |
14:54.04 | Chuji | Cool, how's it coming? |
14:54.09 | dwmw2 | but there's something wrong with the incoming audio still |
14:54.13 | dwmw2 | it's very bizarre. |
14:54.21 | Mother_ | what bluetooth hardware are you using? |
14:54.23 | Chuji | Groovy, glad it's getting some attention |
14:54.49 | dwmw2 | there's distortion I can't quite characterise, and an echo whose delay is proportional to the size of the ring buffer we use between the sco_thread() and blt_read/blt_write |
14:55.01 | dwmw2 | if I feed frames directly from blt_read to blt_write I get perfect echo |
14:55.08 | dwmw2 | using a Plantronics M3000 |
14:55.18 | Mother_ | ok |
14:55.18 | dwmw2 | and an Ericsson T630 |
14:55.35 | dwmw2 | the mailing list seems dead |
14:57.11 | *** join/#asterisk point (1000@213.27.44.55) |
14:57.27 | *** join/#asterisk yaboo (~jsirucka@220.245.131.131) |
14:58.11 | dwmw2 | I'm utterly confused by the echo |
14:58.38 | dwmw2 | I've fixed it to memset the contents of the ring buffer to zero after reading them out. We really shouldn't be hearing them again |
14:58.46 | cypromis | ECHO ECHo ECho Echo echo |
14:59.06 | dwmw2 | <PROTECTED> |
15:02.57 | Zeeek | anyone have a good hardware recommendation for a small quite asterisk box? |
15:07.03 | mmlj4 | there's that linksys wireless thingie, Zeeek |
15:07.11 | mmlj4 | small and cheap |
15:07.23 | dwmw2 | that'd be cute. Is it little-endian? |
15:07.32 | dwmw2 | asterisk is, erm, 'fun' on big-endian machines ATM. |
15:07.35 | mmlj4 | no idea |
15:07.43 | dwmw2 | I'm running it on my powerbook. |
15:07.48 | Zeeek | the WRT54g ? |
15:07.48 | Othello | ... |
15:08.01 | Othello | it's more "fun" to run it with a console driver on a ES1371 |
15:08.11 | ManxPower | "Does this software make my end look big?" |
15:08.18 | mmlj4 | heh |
15:08.19 | Othello | >.<; |
15:09.13 | robl^ | ManxPower: BIG is relative :) |
15:09.26 | *** join/#asterisk zoa (zoa@82.103.76.147) |
15:09.27 | zoa | yo |
15:10.20 | blitzrage | zoa: y0! |
15:10.40 | Damin | zoa: What up? |
15:10.54 | file[laptop] | eek |
15:10.54 | blitzrage | ha! |
15:11.42 | Damin | You know.. as hacked up and ugly as chan_sip is, it has to be the most compatible SIP stack out there. ;) |
15:11.56 | file[laptop] | Damin: ...nah |
15:12.18 | Damin | file[laptop]: According to the guy from SIPFoundry it is.. |
15:12.54 | blitzrage | robl^: lol |
15:13.04 | file[laptop] | if chan_sip is the most compatible SIP stack... something is really really wrong |
15:13.27 | robl^ | its the most compatible, just nothing else is compatible with chan_sip |
15:13.50 | Damin | file[laptop]: As far as communicating with all the other broken shot out there? How many hacks have gone into chan_sip to make it work with other fucked up stacks? |
15:14.08 | file[laptop] | SIP is so, fun |
15:14.34 | zoa | AHA! |
15:14.57 | Damin | file[laptop]: I know people that have gateways that won't communicate with each other, so they bridge them through Asterisk. |
15:15.06 | file[laptop] | Damin: but yes, I agree with that |
15:16.13 | Damin | think for a moment, what would happen if we ripped out the current chan_sip and slapped some other stack in.. |
15:16.44 | file[laptop] | some are too strict, some are too lenient... *sigh* |
15:16.58 | Damin | Within a couple of weeks, the new stack would barely resemble what was originally inserted as it would get hacked up the same way that chan_sip is today just to mainain the compatability that Asterisk has today. |
15:17.11 | ManxPower | ~docs |
15:17.12 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk, or http://www.asteriskdocs.org |
15:17.13 | ManxPower | ~mailinglist |
15:17.14 | jbot | it has been said that mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ |
15:17.42 | Damin | Assuming that is true, aren't we better off sticking with chan_sip and just cleaning it up? |
15:18.12 | file[laptop] | cleaning it up... hehe |
15:18.14 | robl^ | okie. I am off to find some hard drives. never have enufff |
15:19.08 | Damin | robl^: Look under the couch cushions.. I once found a 2 gig laptop drive there.. :) |
15:19.39 | robl^ | Damin: hahah! are you sure you're not my roommate? |
15:20.10 | Mother_ | I really like when it's said that a stack is strict or lenient, if there's a protocol/document/RFC/whatever, *that* is what should be implemented |
15:20.29 | Mother_ | then come Cisco, Nortel, etc. etc. and make their own little 'tweaks' |
15:20.46 | file[laptop] | oh, it's not that |
15:20.48 | ManxPower | There IS chan_sip2.c ya know. |
15:20.57 | file[laptop] | it's because the RFC contradicts itself and stuff... it's hard to implement strict to the RFC |
15:21.22 | robl^ | Damin: I found a Sun Sparchstation IPX, and a mouse for an Atari TT030 under the couch before. |
15:21.30 | file[laptop] | phasing out while reading the RFC is recommended so as to not damage your brain |
15:21.31 | Mother_ | file[laptop]: agreed, but still, there are some things that are blatant |
15:21.37 | Mother_ | LOL agreed too |
15:22.24 | Damin | Mother_: Yeah, but blatant to you, is totally different than blatant to a Cisco engineer. ;) |
15:22.33 | Mother_ | would there be interest in a SIP-to-IAX protocol converter? I'm cooking up something that should be a small box that does protocol translation |
15:22.35 | Damin | Mother_: Sip is a pile of shit protocol.. |
15:22.51 | Damin | Mother_: Well.. that's called "Asterisk".. ;) |
15:23.05 | Mother_ | well yeah, but I would do it the size of half a pack of ciggies |
15:23.32 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
15:23.34 | jontow | so a little smaller than an IAXy to do the job of an IAXy but instead plugging in a SIP phone? |
15:23.40 | jontow | you're scaring me :/ |
15:23.41 | Mother_ | something you could plug a SIP phone into, and get IAX or IAX2 out |
15:23.54 | Damin | Mother_: Hehehe... you are funny.. |
15:23.56 | Mother_ | jontow: yes, along those lines |
15:24.02 | Damin | Mother_: You funny man.. |
15:24.08 | robl^ | Mother_: I already did that.. with a router that runs embedded Linux. Just added a slimmed down copy of Asterisk and a few config files.. does nothing more than convert IAX2 <-> SIP |
15:24.11 | Mother_ | Damin: why? I'm not wanting to do anything to the RTP |
15:24.18 | Blackvel | how can I do this with AGI: "EXEC DIAL ZAP/g1/90,30,tr\n"? |
15:24.27 | Mother_ | robl^: nice |
15:25.01 | Mother_ | my idea is to simply convert the "signalling" so-to-speak, and leave the RTP format alone, whatever codec etc. you may want to use would be irrelevant |
15:25.50 | Mother_ | robl^: what router would that be? |
15:26.21 | Damin | Mother_: WRTG54GS |
15:26.31 | robl^ | ahh. mine is basically to help overcome the SIP/NAT nasties :) It allows someone to use a SIP device inside of NAT and IAX2 out to another NATed Asterisk server |
15:26.45 | robl^ | Mother_: Linksys WRT54 |
15:26.58 | Mother_ | robl^, Damin: thanks, will look into it |
15:27.12 | Damin | Mother_: You obviously don't know Brian Capouche.. |
15:27.33 | Mother_ | who would that be? |
15:28.05 | Mother_ | I get ZGR so it cannot be that well known :) |
15:28.30 | *** join/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
15:28.55 | Mother_ | BTW ZRG == Zero Google Results just FYI |
15:29.10 | MuppetMaster | Hello. |
15:29.13 | Damin | Mother_: http://www.voip-info.org/wiki-Asterisk+Linksys+WRT54G |
15:29.19 | MuppetMaster | Anyone know how to call an xmlrpc from a dialplan? |
15:31.07 | Mother_ | Damin: you appended an 'e' that's why I didn't get anything...I remember playing with shells on a WiFi AP some time ago |
15:31.09 | Mother_ | thanks |
15:31.21 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
15:31.29 | Aze` | re |
15:38.44 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
15:39.22 | *** join/#asterisk bonez39 (~aint@drjones.dsl.xmission.com) |
15:39.37 | dmccollum | Anyone here that understands Festival/mbrola well enough to help point me in the right direction? |
15:40.58 | ManxPower | dmccollum: I have you links to set it up last night. |
15:41.04 | ManxPower | It's not easy. |
15:41.09 | *** part/#asterisk MuppetMaster (~MuppetMas@a82-92-73-185.adsl.xs4all.nl) |
15:43.29 | *** join/#asterisk jeffik (~jeffik@69.158.33.127) |
15:44.28 | *** join/#asterisk mhnoyes (~mhnoyes@user-2ivfmoh.dialup.mindspring.com) |
15:44.54 | *** part/#asterisk logicalonline (~logicalon@border.logicalonline.com) |
15:45.35 | Blackvel | how can I do this with AGI: "EXEC DIAL ZAP/g1/90,30,tr\n"? |
15:47.57 | dmccollum | ManxPower: I looked at all the sites you provided. I installed the mbrola files and made the changes in the config files, but festival complains it needs .scm files that are not part of the mbrola files. I've compared them to the kal_diphone directories and they are no where the same. |
15:50.07 | *** join/#asterisk loick (~loick@APuteaux-151-1-22-92.w82-124.abo.wanadoo.fr) |
15:51.28 | *** join/#asterisk Dabba (~d@ipv6.mfnx.ip6net.net) |
15:52.14 | *** join/#asterisk Jovu (~bert@ev6.net) |
15:53.22 | Dabba | if i do SetVar: var2=1234 in a sample.call file can i use it like Dial(IAX2/8444444@voiptelco/$var2) |
16:00.08 | Jovu | asterisk appears to shut itself down immediately, one minute it loads chan_iax2.so the next its "Beginning asterisk shutdown" for no aparrent reason.. any ideas? |
16:00.58 | cjk | how cpu intensive is sip to iax and iax to sip translation |
16:01.10 | Mother_ | grrrrr |
16:01.43 | *** join/#asterisk coppice (~chatzilla@186.168.17.210.dyn.pacific.net.hk) |
16:01.45 | Mother_ | this may sound silly, but how do I turn off this yellowish highlighting in vim? I pressed some wrong key combination, and it now highlights every # in yellow |
16:01.54 | Mother_ | google is not being helpful today |
16:02.16 | Mother_ | and the docs are too damn long for something this simple, I tried :syn off to no avail |
16:04.09 | CoolAcid | Yahoo? |
16:04.56 | *** part/#asterisk Sebbbb (~sebastian@oven.f0o.de) |
16:05.11 | Jovu | is there any way to find out what's triggering the shutdown? theres nothing i can see in the logs.. |
16:08.24 | Mother_ | nm found it |
16:08.38 | Mother_ | erased the higlighting part out of vimrc |
16:13.55 | *** join/#asterisk Maxxed (Maxxed@pppte03-324.ght.iadfw.net) |
16:14.06 | *** part/#asterisk point (1000@213.27.44.55) |
16:14.44 | Maxxed | should i be worried about a few warnings on asterisks startup ? |
16:15.03 | Maxxed | Mar 26 04:18:19 WARNING[1533]: chan_mgcp.c:4044 reload_config: Unable to get our IP address, MGCP disabled |
16:15.13 | Maxxed | Mar 26 04:18:19 WARNING[1533]: chan_skinny.c:2584 reload_config: Unable to get our IP address, Skinny disabled |
16:15.20 | Maxxed | Mar 26 04:18:19 WARNING[1533]: chan_oss.c:434 soundcard_init: Unable to open /dev/dsp: No such device or address |
16:15.26 | Maxxed | Mar 26 04:18:19 WARNING[1533]: chan_zap.c:9615 setup_zap: Ignoring switchtype |
16:17.09 | Maxxed | the last one kinda looks important |
16:17.21 | Maxxed | oh the machine is just dhcp for testing and what not |
16:17.30 | *** join/#asterisk loud (~ariel@201.139.192.101) |
16:17.37 | Maxxed | it has an ip, so those top few, um, *shurgs* |
16:17.58 | file[laptop] | MGCP is cute |
16:18.12 | Maxxed | yeah i can give a rats about mgcp |
16:18.16 | Maxxed | im a skinny guy :p |
16:18.27 | Maxxed | but none the less, whats with those? |
16:18.43 | file[laptop] | it couldn't get the IP, so they're disabled... duh |
16:18.53 | Maxxed | i dont have a soundcard in the machine, why is it bitchin at me about /dev/dsp |
16:18.57 | file[laptop] | if you wanna use them, set the ip in their config file |
16:19.06 | file[laptop] | because it tried to load a console driver to use a soundcard, and it couldn't |
16:19.09 | Maxxed | file[laptop]: i figured the disabled part ;) |
16:19.22 | Maxxed | file[laptop]: ah, the config, got cha :) |
16:19.43 | Maxxed | file[laptop]: so the soundcard thing, thats not a must is it? |
16:19.55 | file[laptop] | nope... |
16:20.07 | Blackvel | how can i pass EXEC DIAL ZAP/g1/90 inside AGI? |
16:20.22 | Maxxed | warnings n such upset me, red things flashing on my console.. |
16:20.24 | Blackvel | AGI Script Executing Application: (DIAL) Options: (ZAP/g1/90,30,tr) |
16:20.24 | Blackvel | <PROTECTED> |
16:20.25 | file[laptop] | it doesn't matter... you don't need the driver |
16:20.25 | Maxxed | heh |
16:20.32 | Blackvel | that is probably wrong |
16:20.34 | file[laptop] | Blackvel: | instead of , |
16:20.57 | Blackvel | Could not find application (DIAL,ZAP/g1/90,30,tr) |
16:21.00 | Blackvel | oh |
16:21.06 | Blackvel | like EXEC DIAL |
16:21.08 | Blackvel | and then? |
16:21.23 | Blackvel | EXEC DIAL|ZAP? |
16:21.29 | Blackvel | or EXEC DIAL ZAP? |
16:21.32 | file[laptop] | same as you were doing, except for arguments after the Zap use | |
16:21.43 | Blackvel | do I have to pass ZAP/g1/90,30,tr as the AGI option field? |
16:21.54 | file[laptop] | pass ZAP/g1/90|30|tr |
16:24.25 | Blackvel | EXEC DIAL ZAP/g1/90|30|tr\n |
16:24.27 | Blackvel | has to be |
16:24.32 | Blackvel | thnx file :) |
16:24.42 | file[laptop] | uh huh |
16:24.43 | Blackvel | my first xlite->zap call initiated from java |
16:24.44 | Blackvel | hehe |
16:24.52 | file[laptop] | java, that makes me sad |
16:26.25 | *** join/#asterisk zwhitley (~zwhitley@69-162-31-243.stcgpa.adelphia.net) |
16:28.09 | Maxxed | hey file[laptop], u coverd the ip issues, sure enuff i overlooked it in the configs, but what about the....Mar 26 04:18:19 WARNING[1533]: chan_zap.c:9615 setup_zap: Ignoring switchtype |
16:28.26 | *** join/#asterisk bah (048830696@AC92FC1B.ipt.aol.com) |
16:28.30 | *** part/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
16:28.56 | newl | Will an exten using '.' alone be executed regardless if it's been included? i.e. exten => .,1,Setvar(moo=cow) |
16:30.05 | *** join/#asterisk LarsAC (~chatzilla@p508A0A6E.dip0.t-ipconnect.de) |
16:30.46 | Maxxed | pssh.. me again jacking off in my configs.. |
16:30.47 | Maxxed | switchtype=national |
16:31.30 | Maxxed | so! im good, im down to one warning |
16:31.40 | Maxxed | and thats the soundcard /dev/dsp one... |
16:31.49 | Maxxed | in what config should i look to disable that |
16:31.57 | file[laptop] | just don't load the module |
16:32.02 | file[laptop] | noload => chan_oss.so in modules.conf |
16:32.07 | Maxxed | so in the modules.conf |
16:32.10 | Maxxed | ah thanks :) |
16:32.24 | Maxxed | i am a noob, so thanks for bearing with me :p |
16:33.06 | Maxxed | ;noload => chan_alsa.so |
16:33.06 | Maxxed | ;noload => chan_oss.so |
16:33.12 | Zaw | i'm thinking about ditching livevoip.com since their whole local DID fiasco. anyone have a recommended alternative for IAX? |
16:33.42 | LarsAC | what format is the value of the origtime tag in the .txt voicebox files ? |
16:35.00 | Maxxed | success! |
16:35.06 | Maxxed | mr file[laptop], i ow u a soda ;) |
16:35.42 | file[laptop] | yeh yeah |
16:36.04 | Maxxed | :) |
16:36.27 | file[laptop] | paypal donations accepted :p |
16:36.53 | file[laptop] | haha |
16:37.11 | Maxxed | :) |
16:37.31 | file[laptop] | joshnet@nbnet.nb.ca everyone - donate a buck - get a truck - run amuck - oh just die :p |
16:37.45 | Maxxed | heh |
16:38.15 | file[laptop] | if you happened to have a TV tuner card that is Linux compatible though, I'd accept that too |
16:39.26 | Maxxed | You have sent cash! An email has been sent to the recipient. |
16:39.29 | newl | any bt8x8 card should do the trick. :) |
16:39.37 | file[laptop] | pure silly |
16:39.44 | Mother_ | notlame rocks |
16:39.49 | file[laptop] | yes yes |
16:39.55 | file[laptop] | but if anyone has one they'd give me, I'd like that :p |
16:40.02 | Maxxed | um, i gota few ati's somewere er unuther, but i dobt there linux compat, hell there bearly windows compatable |
16:40.08 | Maxxed | heh im silly like putty |
16:40.17 | file[laptop] | all in wonders? |
16:40.28 | Maxxed | nutty like squirrel turds |
16:40.33 | *** join/#asterisk bonez39 (~aint@drjones.dsl.xmission.com) |
16:40.46 | Maxxed | um, there like, value tv or some bs |
16:40.52 | file[laptop] | creepy |
16:40.55 | Maxxed | ur best bet is ebay :p |
16:41.02 | file[laptop] | yeah yeah |
16:41.17 | file[laptop] | Maxxed: thx regardless |
16:41.21 | Maxxed | heh |
16:41.26 | Maxxed | np :) |
16:42.12 | *** join/#asterisk E|nyPRI_ (les@S0106000bdb97681e.wp.shawcable.net) |
16:42.12 | *** join/#asterisk gopherspidey (~spidey@12-216-167-110.client.mchsi.com) |
16:42.20 | E|nyPRI_ | Hi |
16:42.30 | gopherspidey | Hi |
16:42.45 | E|nyPRI_ | I have a problem with my *. When I receive a call to it, it runs the context twice. |
16:43.09 | E|nyPRI_ | http://pastebin.ca/8279 |
16:43.37 | E|nyPRI_ | so i head "Invalid extension" twice. |
16:44.19 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
16:44.25 | gopherspidey | It plays invalid extension twice then hangsup |
16:44.26 | Maxxed | um, mabe comment out the ball sack in there ;p |
16:44.27 | Maxxed | har har har.. |
16:44.44 | file[laptop] | don't... don't... use _. |
16:44.55 | Maxxed | use s |
16:45.01 | file[laptop] | or _X. if you HAVE to |
16:45.22 | E|nyPRI_ | gopher; yes. |
16:45.59 | gopherspidey | E|nyPRI_: use the "s" extension |
16:46.05 | E|nyPRI_ | whoah. ok, I used _X. |
16:46.07 | E|nyPRI_ | and it worked |
16:46.18 | E|nyPRI_ | dunno why that did not occure to me |
16:46.31 | Maxxed | E|nyPRI_, file does take paypal donations :p |
16:46.40 | file[laptop] | yes, I do |
16:46.47 | E|nyPRI_ | :) |
16:46.50 | file[laptop] | crazy coconuts |
16:46.57 | Maxxed | :p |
16:47.07 | *** part/#asterisk gopherspidey (~spidey@12-216-167-110.client.mchsi.com) |
16:47.13 | file[laptop] | I can actually get two sodas with that at the corner store |
16:47.16 | *** join/#asterisk gopherspidey (~spidey@12-216-167-110.client.mchsi.com) |
16:47.34 | Maxxed | oh well, i have a lil assistance credit :p |
16:48.48 | Moc | hi E|nyPRI_ |
16:49.43 | gopherspidey | Does anyone know if someone is working on the new Asterisk Manager interface? |
16:49.47 | gopherspidey | http://www.voip-forum.com/news.php?p=168&c=1 |
16:51.43 | *** join/#asterisk IQ (~IQ@70-59-165-40.omah.qwest.net) |
16:53.33 | *** join/#asterisk Sedorox (~Sed@Neptune-W.client.wlgrv.pa.sed6.net) |
17:02.22 | *** join/#asterisk lilneon (~tj_r3@cuscon12874.tstt.net.tt) |
17:02.30 | lilneon | hey good mornin guys |
17:03.14 | Maxxed | "Groundstart signalling is sometimes used by PBX's. If you don't know what it is, don't worry, you won't need it." |
17:03.16 | Maxxed | lol |
17:04.06 | *** join/#asterisk MikeJ[Jayden] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
17:04.26 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
17:04.26 | *** mode/#asterisk [+o bkw_] by ChanServ |
17:05.29 | Maxxed | well im off for now |
17:05.38 | Maxxed | prob be back in a bit, ima trash my * configs :p |
17:05.45 | Maxxed | harass yous guys lata ;) |
17:05.49 | *** join/#asterisk mitmit (~mitmit@CPE000c418b4787-CM001225401fee.cpe.net.cable.rogers.com) |
17:05.51 | Maxxed | oh and thanks again file[laptop] :D |
17:06.21 | Jovu | my asterisk server begins starting up, and then appears to shut itself down for no aparrent reason.. the logs aren't very helpfull either.. anyone care to take a look? |
17:08.43 | E|nyPRI_ | moc; hi |
17:09.27 | E|nyPRI_ | start it from the command line using asterisk -vvvvvvvvvvvvvvvvvvvvvvcg |
17:09.34 | E|nyPRI_ | see where it bails |
17:09.38 | *** join/#asterisk lilneon_ (~tj_r3@cuscon12874.tstt.net.tt) |
17:09.46 | lilneon_ | hey everyone mornin again.. |
17:09.55 | *** join/#asterisk goobster (goobster@c-67-168-105-166.client.comcast.net) |
17:09.56 | lilneon_ | anyone here? |
17:10.03 | IQ | morning |
17:10.10 | cactus1_away | good morning |
17:10.12 | *** join/#asterisk CaptChris (~Chris@c-67-181-99-1.client.comcast.net) |
17:10.35 | lilneon_ | hey guys, if i wanted to connect to my local telco's gsm network.. what would be the best method? |
17:10.45 | lilneon_ | that is connect my asterisk box to the gsm network |
17:10.49 | Moc | by gsm you mean cellular É |
17:11.02 | lilneon_ | Moc: yeah |
17:11.07 | Jovu | E|nyPRI_, i tried that... doesnt seem to be very helpfull, can i paste the last few lines in /query ? |
17:11.14 | E|nyPRI_ | run a DS1 to their access tandem |
17:11.40 | E|nyPRI_ | jovu; pastebin.ca |
17:11.42 | lilneon_ | and if they aren't willling? |
17:11.46 | DyOS | hey does anyone have any recommendations as to what is the best cheapest voip provider to use for international calling unlimited |
17:11.52 | Moc | yes, just put a DS1 to them... or if your cheap, you can get those cellphone docking station, but those are crap hehe |
17:11.56 | E|nyPRI_ | You in canada or usa ? |
17:12.25 | cactus1 | whats a DS1? |
17:12.27 | Jovu | the output seems to be different every time |
17:12.27 | E|nyPRI_ | t1 |
17:12.46 | lilneon_ | cactus1:googling it now... gooogle is your friend :D |
17:12.48 | E|nyPRI_ | cuz if your in Canada/USA they cannoy refuse you equal access. |
17:12.57 | E|nyPRI_ | dunno how it works otherplaces |
17:12.58 | gopherspidey | 1.5M or 24 ds0 |
17:13.01 | cactus1 | indeed lilneon |
17:13.12 | Moc | E|nyPRI_, they can't, but they can charge you alot I guess ;) |
17:13.19 | lilneon_ | E|nyPRI: well i am not in the US.. live in the caribbean |
17:13.28 | E|nyPRI_ | Oh. dunno how it is regulated there |
17:13.41 | lilneon_ | country has like one or two telco's who abusing their monopoloy big time!!! |
17:13.42 | E|nyPRI_ | moc; all those rates are tariff'd. you get charged the same as every other carrier. its not dependant on volume. |
17:13.49 | cactus1 | carribean thats friggin sweet :) |
17:14.00 | Jovu | E|nyPRI_, http://pastebin.ca/8281 |
17:14.12 | Moc | E|nyPRI_, I dont even know what exactly connecting to the 'gsm' network do .. |
17:14.15 | CaptChris | hello all. i'm looking for some help in setup/config. |
17:14.24 | Moc | free outgoing and incoming call to cell phone for you |
17:14.25 | Moc | ? |
17:14.43 | CaptChris | i'm using a Mac, and i think i have everything setup proper.... |
17:14.58 | E|nyPRI_ | unknown. cellular networks are a bit of a mystery to me. |
17:15.16 | E|nyPRI_ | you'd probably be able to terminate calls cheaper. |
17:15.18 | CaptChris | while using Jon's Phone Tool trying to connect to vm extension (or any extension) ... |
17:15.39 | Moc | I wish I found a cheap PRI provider in mtl.. |
17:15.43 | E|nyPRI_ | jovu; hm dunno! |
17:15.45 | CaptChris | I keep getting error message of"channel.c:1817 __ast_request_and_dial: Unable to request channel SIP" |
17:15.50 | E|nyPRI_ | moc; whats the price for pri there? |
17:16.01 | IQ | E|nyPRI_, Moc: Free SMS ? |
17:16.03 | E|nyPRI_ | jovu; did you just compile it? or is that a binary version |
17:16.14 | Moc | well cheapest I found is 650$ for ye year contract... but I can't get that price rightnow |
17:16.22 | Moc | ye = 1 |
17:16.30 | Moc | it about 750 to 1000$ |
17:17.08 | Jovu | E|nyPRI_, its compiled from cvs, i was using 1.0.6 and 1.0.7 before with the same issues.. i've no idea where its failing either, the output is often different depending how far each thread got before it got the shutdown signal |
17:17.14 | CaptChris | can anyone help? I've tried google for the past week, but no luck |
17:18.38 | E|nyPRI_ | moc; $750 is reasonable |
17:18.47 | bparker | who is the best voip provider (with IAX access)? |
17:18.50 | Sedorox | CaptChris: have you restarted asterisk? |
17:19.02 | Moc | that still 32$ per channel |
17:19.03 | CaptChris | yes... and reload several times |
17:19.19 | E|nyPRI_ | moc; yep. PRI isnt much cheaper than pots |
17:19.29 | cactus1 | bparker: voicepulse is good if your in the USA |
17:19.39 | E|nyPRI_ | unless your colocated with the telco. ie: Colo with GT here in Winnipeg, PRI's are $600 or $490 for incoming only |
17:19.41 | Mother_ | question: on a large Siemens PBX I worked with some time ago there was a function that allowed you to get a callback when someone that you called was busy became available, by pressing a key combination, is there an equivalent in *? |
17:20.00 | Moc | E|nyPRI_, incoming only doesnt give you callerid + name ? |
17:20.01 | E|nyPRI_ | moc; go with a clec tho. because they can negotiate price. versus ilec bell who cannot |
17:20.09 | E|nyPRI_ | moc; sure it does. |
17:20.19 | *** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc) |
17:20.22 | Moc | btw how do you manage with no NI3 support in * ? |
17:20.30 | E|nyPRI_ | NI3 ? |
17:20.43 | gopherspidey | Do any one know if someone is working at "http://www.voip-forum.com/news.php?p=168&c=1"? |
17:20.49 | E|nyPRI_ | by incoming only, I mean its a normal PRI, but you cant make outbound calls |
17:20.56 | Moc | well from a GT doc, they support Caller Name only with NI3 support (maybe) |
17:20.57 | E|nyPRI_ | ISP PRI is what they're called sometimes |
17:21.06 | Moc | E|nyPRI_, the PDF Ive read doesnt have it, hold |
17:21.11 | E|nyPRI_ | Oh. gt yeah, that propogandah is a bit off. |
17:21.11 | newl | NI3, kinda like NA3, only for ISDN. :D |
17:21.48 | E|nyPRI_ | it is unlikely you will find a telco PRI provider who cannot send you callerid/name in Canada. |
17:21.49 | Moc | E|nyPRI_: http://www.gt.ca/pdfs/products/Digital_GT_EN.pdf |
17:22.04 | Sedorox | CaptChris: I think something is wrong in either youe extentions.conf or sip.conf |
17:22.09 | E|nyPRI_ | yeah, i've read that exact same pdf. |
17:22.14 | Moc | pri isp has no did |
17:22.30 | E|nyPRI_ | well, it can. |
17:22.41 | Moc | damn GT!! |
17:22.42 | *** join/#asterisk PTG123 (~PTG@ip68-106-24-139.ph.ph.cox.net) |
17:22.42 | E|nyPRI_ | just have to ask for it |
17:22.49 | PTG123 | howdy ho |
17:23.05 | *** join/#asterisk lilneon (~tj_r3@cuscon14967.tstt.net.tt) |
17:23.14 | lilneon | hi again.. |
17:23.22 | E|nyPRI_ | I've asked them to turn off CallerID screening :) (ie: 100% spoofable outbound CallerID) |
17:23.26 | lilneon | dumb internet keeps disconnecting here.. |
17:23.27 | Moc | ok, anyway, I gota find a buisness who need a PRI first.. |
17:23.35 | Moc | E|nyPRI_ good |
17:23.46 | CaptChris | Sedorox: I would guess so as well. i'm very new to Asterisk and have only SIP software currently. Also haven't been able to establish any connections... but I can tell that I the SIP client is talking to Asterisk (via logs) |
17:24.09 | Sedorox | you using a soft phone? |
17:24.10 | Moc | E|nyPRI_, Broadvox doesn't have their caller id blocked |
17:24.24 | E|nyPRI_ | is broadvox canadian? |
17:24.26 | CaptChris | yes "Jon's Phone Tool" mostly |
17:24.33 | CaptChris | sometimes X-Lite |
17:24.35 | Sedorox | never heard of it... anyway |
17:24.38 | Moc | E|nyPRI_ they have canadian server with GT |
17:24.44 | Moc | I had a few cnd DID with them |
17:24.59 | Sedorox | pastebin me your extentions.conf.... the part that your trying to dial |
17:25.01 | Moc | but broadvox su..k |
17:25.02 | E|nyPRI_ | yeah. by default callerid screening is on. you have to issue a NSR to have it removed, and then deal with their legal department. |
17:25.19 | Moc | yea |
17:25.48 | Moc | well if Im really hired by this other buisnes, I'll do * deployment in mtl fulltime and developpement also .. |
17:26.01 | CoolAcid | Anyone written some interesting dialplans as of yet? Todo something usless or anything? |
17:26.08 | Sedorox | Mother_: You could do that.. I don't think there is anything in Asterisk now for it... but I would think it would combine macro's.. and maybe agi's |
17:26.10 | CoolAcid | s/usless/useless |
17:26.18 | CaptChris | Pastebin? .. is that a DCC Send? |
17:26.24 | Sedorox | ~pastebin |
17:26.25 | jbot | methinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
17:26.28 | Moc | so I'll probably have budget for a PRI!!! Can't wait |
17:26.38 | E|nyPRI_ | :-D |
17:26.47 | E|nyPRI_ | just wait till you find out that the 800# called is not delivered :P |
17:27.03 | Moc | ? |
17:27.07 | E|nyPRI_ | so you'll need a normal DID for every 800 |
17:27.11 | *** join/#asterisk lilneon (~tj_r3@cuscon14609.tstt.net.tt) |
17:27.14 | lilneon | k back |
17:27.21 | implicit | E|nyPRI_, by 100% do you mean any #? i wasn't aware that they allow for intl cid's to be set through them even after you ask them to remove |
17:27.22 | lilneon | anyone got my last question? |
17:27.45 | Sedorox | CaptChris: I can't do dcc's |
17:28.01 | Sedorox | just copy and paste the content into pastebin.ca |
17:28.12 | E|nyPRI_ | implicit; as far as I know yes. I've talked with a bunch of their switch techs & isdn guru's. basically, I said I want to be able to set my caller ID to anything. they said it can be done, just need a non-standard request, and then legal dept. |
17:28.15 | lilneon | guys |
17:28.18 | Sedorox | lilneon: doesn't look like it |
17:28.46 | lilneon | ok.. what are my options if a DS1 not a possibility? |
17:28.55 | lilneon | need to connect to a cellular network.. |
17:28.58 | Moc | E|nyPRI_, you can set the caller ID & Name with GT right ? |
17:29.04 | IQ | Hi, any h.323 softphone? |
17:29.09 | Sedorox | dunno here... |
17:29.14 | implicit | E|nyPRI_, usually they might think you are talking about +1 CID |
17:29.15 | E|nyPRI_ | moc; yes. |
17:29.22 | Moc | I love canada for that .. |
17:29.34 | implicit | Moc, yep |
17:29.45 | CaptChris | Sedorox:excuse my ignorance. It's also been a long time since I've used irc. I'm not familiar with pastebin. can you clarify? |
17:29.47 | E|nyPRI_ | my caller name has not worked in a while thouhg, not sure where the problem exists. It worked in 1.0.5 but 'appears' to of stopped working in 1.0.6+ and CVS head |
17:29.52 | Moc | E|nyPRI_, you connect * with GT and everything is great ? |
17:29.57 | Sedorox | goto http://www.pastebin.ca |
17:29.58 | CoolAcid | CaptChris: Goto www.patebin.ca |
17:30.03 | implicit | not patebin.ca |
17:30.07 | implicit | it is a porn site |
17:30.10 | implicit | just pastebin.ca |
17:30.13 | CaptChris | ok |
17:30.15 | CoolAcid | arg.. mist the s.. my bad.. |
17:30.17 | Sedorox | yea.. ignore CoolAcid :-p |
17:30.18 | E|nyPRI_ | well, i use GT. It works, its been reliable. But I've had a few support issues which resulted in my crying to the CRTC |
17:30.30 | Sedorox | and you can paste large blocks of text |
17:30.37 | Moc | gt is bell now so ... :( |
17:30.38 | Sedorox | then when you submit... give me the URL |
17:30.49 | E|nyPRI_ | yeah, but only in ownership, the network has not been integrated with them. |
17:30.56 | E|nyPRI_ | so for all intents and purposes, gt <> bell. |
17:31.03 | E|nyPRI_ | for now. |
17:31.16 | Moc | yea, well I'll continue to dig up a PRI somewhere |
17:31.23 | E|nyPRI_ | ok, gotta go to the chiro. and let him kick my ass. ttyl. |
17:31.23 | file[laptop] | dig up a PRI... haha |
17:31.28 | file[laptop] | I could imagine you literally doing that |
17:31.34 | Moc | hehe |
17:31.37 | file[laptop] | going outside, finding someone's PRI, and moving it to your place |
17:31.45 | IQ | H.323 SoftPhone - anyone ? help ? |
17:31.55 | Moc | IQ, good luck is all I could say |
17:32.10 | IQ | Moc: thanks :P |
17:32.17 | Sedorox | whats so good about H323? |
17:32.26 | Moc | nothing is good about h323 |
17:32.27 | file[laptop] | absolutely nothin' |
17:32.29 | Sedorox | lol |
17:32.29 | IQ | Got Avaya at work :( |
17:32.35 | Sedorox | then why do so many people use it? |
17:32.40 | Sedorox | thats what they use? |
17:33.09 | Moc | Sedorox, is sadly was the first out, in today world, only video conference use h323, and outdated voip software |
17:33.12 | CoolAcid | IQ: http://www.openh323.org/ Openphone and OhPhone |
17:33.16 | CaptChris | Sedorox: OK. I think I have it http://pastebin.ca/8283 |
17:33.17 | IQ | got to do some debugging - and dont want to use Avaya's IP SoftPhone |
17:33.22 | Sedorox | Moc: gotcha |
17:33.32 | *** join/#asterisk G0shen (~Goshen@70-57-80-147.slkc.qwest.net) |
17:33.36 | Mw3 | anyone knows vpn capable device with 1or2 fxs port? |
17:33.38 | file[laptop] | yeah my school uses H323 |
17:33.42 | Sedorox | CaptChris: ok.. now.. what extention you trying to dial? |
17:33.54 | CaptChris | Sedorox: OK. I think I have it http://pastebin.ca/8283 |
17:34.03 | Sedorox | CaptChris: yes.. I have it loaded |
17:34.06 | Syncros | linux box + pci fxs |
17:34.09 | Sedorox | what extention are you trying to dial? |
17:34.09 | CaptChris | 8500 |
17:34.17 | Sedorox | ok |
17:34.23 | IQ | CoolAcid: was looking more like x-lite :) . openh323.org is a good site tough |
17:34.27 | Sedorox | so trying to dial voicemail.. from a softphone.... |
17:34.52 | CoolAcid | IQ: Sorry dude. You asked for softphones, it's the only place I know that does H323 softphones :) |
17:34.59 | CaptChris | yes |
17:35.12 | Sedorox | CaptChris: ok.... now pastbin the contents of your sip.conf that pertain to the phone in question.. and X out the password(s) |
17:35.13 | IQ | CoolAcid: yeah, u r right - thanks :) |
17:35.37 | CoolAcid | No worries.. Now if only I could figure out whats wrong with my IAXtel :) |
17:35.44 | CaptChris | ok. 2 files will be in pastebin for the complete config |
17:36.07 | Sedorox | ummm ok |
17:38.09 | lilneon | hey, who told me about the docking stations for cellular phones?... what problem did u have with them? |
17:38.26 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.rr.com) |
17:38.27 | *** mode/#asterisk [+o anthm] by ChanServ |
17:38.50 | IQ | lilneon: I think you'll need an ATA with cell phone's dock station |
17:39.11 | Sedorox | IQ: actually.. a FXO |
17:39.13 | CaptChris | Sedorox: got it. 3 files in sequence in the pastbin http://pastebin.ca/8284 |
17:39.29 | Sedorox | CaptChris: is that |
17:39.30 | Sedorox | 301 #include /etc/asterisk/SIP/SIP-incoming.conf |
17:39.31 | Sedorox | 302 #include /etc/asterisk/SIP/SIP-100.conf |
17:39.31 | Sedorox | 303 #include /etc/asterisk/SIP/SIP-200.conf |
17:39.32 | Sedorox | files? |
17:39.40 | IQ | Sedorox: yeah - or ATA that has FXO :) |
17:39.40 | CaptChris | yes |
17:39.45 | Sedorox | IQ: true :-p |
17:39.55 | lilneon | IQ:true.. know anywhere i could find one? apart frm ebay ofcourse |
17:39.55 | blitzrage | does anyone know what iaxcompat actaully does? |
17:40.00 | Sedorox | CaptChris: ok.... 1. what phone you dialing from? |
17:40.24 | IQ | lilneon: they are everywhere... I got SPA-3000, its a great ATA |
17:40.33 | IQ | lilneon: with 1 FXO and 1 FXS |
17:40.45 | Sedorox | and it isn't that expensive either |
17:40.46 | CaptChris | 100 |
17:40.52 | lilneon | k cool |
17:41.06 | Sedorox | CaptChris: do you have anything included in sip.conf? |
17:42.03 | CaptChris | only the 3 files listed at the end |
17:42.14 | Sedorox | in sip.conf? not extentions.conf... |
17:42.57 | CaptChris | yes. it's the same 3 listed in both |
17:43.08 | Sedorox | well I can tell you right no |
17:43.13 | Sedorox | I've never seen a config like that before |
17:43.18 | Sedorox | either something's really messed up |
17:43.26 | Sedorox | or I didn't know you could intermix sip and extentions |
17:43.52 | Sedorox | do yourself a favor |
17:43.56 | CaptChris | ah... perhaps thats the trouble. |
17:44.04 | Sedorox | get rid of the includes till your more farmiliar with Asterisk |
17:44.05 | Sedorox | then... |
17:44.13 | Sedorox | move the extentions stuff into extentions.conf |
17:44.17 | Sedorox | and the sip stuff into sip.conf |
17:44.35 | CaptChris | i can't remember where i had seen this type of config set |
17:44.37 | Sedorox | and you don't need type=friend in the context either :-p |
17:44.40 | CaptChris | ok. i will do that |
17:44.47 | Sedorox | so try that |
17:44.54 | Sedorox | I'll be back on later this afternoon... |
17:45.03 | Sedorox | I g2g... need to fix a computer.. then lunch/dinner |
17:45.35 | Sedorox | so good luck.. but just seperate that stuff out.. make sure sip stuff is only in sip.conf.. and extentions stuff is only in extentions.conf... |
17:45.36 | Sedorox | enjoy |
17:46.08 | IQ | CaptChris: Did you try "make samples" after installnig asterisk? |
17:47.20 | CaptChris | thanx. i have changed the extensions and sip config files. will reload and try again |
17:48.05 | CaptChris | same trouble |
17:48.10 | lilneon | guys, if i am buying a tdm400P card frm digium, going to connect it to a channel bank, or perhaps a straight t1 what config should i take? 4fxs,4fxo's or a mixture.. |
17:48.39 | lilneon | couldn't i take 4fxs and have the channel banks contain fxo's and fxs channels? |
17:49.09 | IQ | CaptChris: its a new installation? |
17:49.21 | Qwell | lilneon: gonna get a tdm for a T1? |
17:49.23 | CaptChris | IQ: yes |
17:49.31 | DyOS | i have a question about voicepulse...say i get the $45/month unlimited....If two sip devices make calls at the same time using the IAX connection what happens? Does one of them get a busy tone or do they both go through? |
17:49.37 | Qwell | You'd need like...6 of them |
17:49.46 | IQ | CaptChris: Did you try "make samples" ? |
17:49.48 | Zeeek | CaptChris can you repeat in two sentences what the trouble is? I missed that part |
17:50.03 | Zeeek | you want to call 8500 from a soft=phone and... what? |
17:50.04 | Qwell | DyOS: I believe they both go through, but it charges more for the second call. |
17:50.10 | Qwell | DyOS: call and ask to make sure |
17:50.12 | IQ | lilneon: why can't they make it configurable - technical limitation? |
17:50.14 | newl | I need to execute an extension block without an extension number. [test] include => moo ; other includes [moo] exten => s,1,Playback(is), the [moo] context never gets ran. Specifying _. runs the extension twice. Ideas? |
17:50.18 | CaptChris | IQ:I tried to compile Asterisk myself, but I can't remember if I had too much trouble and simply installed a package from someone else |
17:50.21 | *** join/#asterisk mbranca (~matteo@host-84-222-14-188.cust-adsl.tiscali.it) |
17:50.49 | CaptChris | IQ: but I'm certain I didn't "make samples" |
17:50.58 | IQ | CaptChris: sorry, I didn't read your old messages. |
17:51.34 | IQ | CaptChris: doing "make samples" give you some default .conf files - which is a good start even if you are an expert (unless you want to copy your.conf files from another machine) |
17:51.56 | Zeeek | CaptChris what happenes on the CLI when you dial? |
17:52.06 | lilneon | IQ,Qwell: yeah they can make it configurable.. i just wanted to know if i should take a quad t1 card (T410P) or a TDM400p with (fxs/fxo).. this still kinda confuses me bout fxs and fxo.. :S |
17:52.30 | Qwell | lilneon: If you're getting a full T1, the T1 card is best |
17:52.33 | CaptChris | there were alot of .conf files after the install... i'm they did appear to be samples |
17:52.41 | *** join/#asterisk PCadach (~paul@www.east.telecom.kz) |
17:53.03 | CaptChris | message of "__ast_request_and_dial: Unable to request channel SIP/100 |
17:53.14 | ManxPower | lilneon: FXO provides dialtone and ring voltage. FXS expects to receive dialtone and receive ring voltage |
17:53.24 | lilneon | Qwell: this config is simply going to be analogphones & POTs->channel bank->asterisk->voip... for now.. but i need to think bout the future.. adn expandability |
17:53.39 | CaptChris | and now getting "No such Host: 100" |
17:53.43 | lilneon | thnx manxpower |
17:53.52 | Qwell | lilneon: well, 6 TDM cards isn't exactly what I would call "expandable" |
17:54.14 | Zeeek | CaptChris here's a suggestion: read the fiirst article below: |
17:54.15 | Zeeek | Starter tutorial: |
17:54.15 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
17:54.15 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
17:54.15 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
17:54.15 | Zeeek | THE reference of the moment: |
17:54.17 | Zeeek | http://www.asteriskdocs.org |
17:54.27 | Zeeek | then you will see what the lines in those files mean |
17:54.37 | Zeeek | it's only a few pages long |
17:54.38 | lilneon | qwell:k cool.. so that settles it... quad t1 then.. thnx guys |
17:55.29 | CaptChris | ok. I will do that. then if I still have troubles...I'll be back... but probably not until late tonight (PST) or tomorrow |
17:55.47 | Zeeek | CaptChris if you read that one article, you'll be miles ahead believe me |
17:55.54 | CaptChris | I think my biggest problem is in not fully understanding SIP and extensions. |
17:55.59 | CaptChris | thanks for the help |
17:56.07 | blitzrage | Asterisk requires actual reading unfortunately... you can't fake it :) |
17:56.19 | Zeeek | exactly: the SIP stuff defines the clients. The extensions.conf defines the dialplan |
17:56.24 | Shido6 | ... |
17:56.30 | Zeeek | blitzrage - I can fake it |
17:56.31 | CaptChris | yes! I seem to have been studying for past 2 weeks now |
17:56.41 | blitzrage | Zeeek: I fake things plenty :) |
17:57.06 | Zeeek | I never got the 70's show |
17:57.16 | CaptChris | now i need to run. thanks again all for the help |
17:57.16 | Zeeek | Too busy reading about asterisk I guess |
17:57.28 | Zeeek | ok CaptChris good luck |
17:57.44 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
17:57.45 | Zeeek | so... family and tree ? |
17:58.07 | Zeeek | <PROTECTED> |
17:59.49 | DyOS | I'm looking at the voiceblue product which provides ip to gsm network connectivity...can someone explain to me why someone would want to do this and pay cellular fees? Why not just use a pstn connection to dial gsm #'s....? |
18:00.29 | IQ | DyOS: link please |
18:00.50 | Zeeek | in this country, it's cheaper to call gsm from gsm than landline |
18:00.55 | Zeeek | maybe that's why? |
18:01.19 | Zeeek | however, there are voip providers that offer cheaper rates |
18:01.19 | DyOS | http://www.2n.cz/products/gsm_gateways/voip/voiceblue.html |
18:02.10 | DyOS | wait you're saying it's cheaper to call gsm from gsm....but i'm looking at this voiceblue device and it looks like it takes sim cards...I'm imaging you could take the sim card outof yoru gsm capable cell phone and throw it in that box |
18:02.28 | DyOS | only problem is...you're going ot incur cellular charges as if you were using your cell phone...i guess i just don't understand |
18:02.37 | DyOS | why would you want ot do this and not just use viop or pstn |
18:02.42 | DyOS | to connect to the gsm network |
18:02.49 | Qwell | DyOS: Don't forget about "IN" or "sprint to sprint" |
18:02.56 | Shido6 | dood |
18:03.09 | Qwell | I just woke up though, and I didn't RTFA |
18:03.14 | IQ | DyOS: I would do that to send/receive SMS all over the world |
18:03.30 | Shido6 | setup your own cell net |
18:03.48 | Qwell | Shido6: lets do it |
18:03.55 | Shido6 | Ive done it in .au |
18:04.03 | IQ | only very few carriers in USA can do international SMS |
18:04.19 | Qwell | Shido6: fairly easy? |
18:04.21 | DyOS | shido my man my spandsp still isn't working |
18:04.23 | newl | Shido6: we use the crows for communication here. :) |
18:04.26 | Qwell | expensive though, I'm sure |
18:04.50 | Shido6 | no |
18:04.52 | Shido6 | wow |
18:05.02 | Shido6 | I cant beleive they FINALLY have instructions on their page |
18:05.14 | Shido6 | that wasnt there when I went through it |
18:05.31 | Qwell | What has instructions? |
18:05.33 | Shido6 | voice and data on one 1 and out another port to the damn thing, struggled for like 3 days |
18:05.43 | Shido6 | http://www.2n.cz/news/press_releases/voiceblue_with_asterisk.html? |
18:05.46 | Qwell | oh |
18:07.56 | DyOS | so i'm not sure i understand yet can someone explain to me the benefit of routing gsm calls when you coudl just use voip or pstn to call cellular? |
18:10.11 | DyOS | you still have to buy the gsm service to get the sim card to stick into the voiceblue device correct? |
18:10.15 | jakepdev | I'm trying to make libpri and I get an error on this line "include .depend" |
18:11.27 | Shido6 | and if you dont have cellular in your area? |
18:11.51 | Shido6 | if you dont want to use any of the super expensive cellular networks that are in the area? |
18:12.03 | Shido6 | $1.05 to call anywhere |
18:12.04 | *** join/#asterisk montoya (montoya@200.195.87.156) |
18:12.12 | Shido6 | per minute |
18:12.40 | jakepdev | I think I found out how to make this Avaya connection work |
18:12.46 | jakepdev | 2b xfer |
18:12.52 | jakepdev | Avaya supports it |
18:13.03 | DyOS | so the point of voiceblue would be to start your own gsm cellular network and charge people to use it then? |
18:13.04 | gopherspidey | Some GSM networks in parts of the world are inexpensive |
18:13.15 | jakepdev | just gotta make the newest build of * |
18:13.38 | Zeeek | DyOS here's one answer: http://www.2n.cz/news/press_releases/case_study_small_office.html |
18:13.52 | lilneon | shido6 : cool idea.. but how does it hook up to the cellular netwrks? u add in sim cards or something .. sorry haven't reaqd thru anything on the site just yet |
18:16.33 | gopherspidey | Yep, just the sim card that the provider gives you |
18:16.56 | cripito | :} there is lot of ppl using this also with stolen sim cards :D |
18:17.02 | lilneon | sweet |
18:17.08 | DyOS | but you're still going ot incur normal cellular costs of whatever sim card you stick in that box correct? |
18:17.21 | cripito | yes. |
18:17.38 | lilneon | Dyos: for me it would wrk great... three cellular companies here.. interconnection between them real high |
18:17.50 | lilneon | i could use this to bypass the interconnection feees |
18:17.55 | cripito | brb |
18:17.59 | DyOS | so then the only ppl that would benefit from this are people where there are no telephone lines high speed internet etc? |
18:18.36 | lilneon | Dyos: well no.. i think u are looking at this the wrong way |
18:18.42 | DyOS | explain to me |
18:19.11 | Zeeek | actaully the benefits are given on the site - maybe read it? |
18:19.20 | lilneon | Dyos: well.. i only now checked this out.. but frm what i can see.. i could use it to connect to cellular networks without using a ds1 |
18:20.11 | lilneon | dyos: usually more expensive to connect frm gsm to landlines and vici versa.. getting them to connect directly would be of some use to me.. and i could also allow them to use VOIP for long distance.. |
18:20.20 | *** join/#asterisk marks_ (~Marks_Dis@cpe-70-112-81-84.austin.res.rr.com) |
18:20.22 | marks_ | Hello? |
18:20.43 | marks_ | Will someone pm me and give me a howto install ASTERISK on GENTOO mini INSTALL.. I WILL PAY!! needs to work with FWD!! |
18:21.39 | IQ | I think mark got too many PMs |
18:21.40 | *** join/#asterisk marks_ (~Marks_Dis@cpe-70-112-81-84.austin.res.rr.com) |
18:21.59 | IQ | marks_: too many PMs ? |
18:22.06 | marks_ | nope |
18:22.12 | marks_ | i closed the connection |
18:22.12 | lilneon | dyos: ideas flowing yet? |
18:22.15 | marks_ | i didnt gget any!! |
18:22.18 | IQ | marks_: I thought ur ISP couldn't handle all PMs u r getting :P |
18:23.09 | DyOS | holy shit i think i just put 2 and 2 together and got 5....so say you have 2 companies in diff locations instead of using the internet to route your calls you just get 2 accounts with a normal gsm phone company that allows internetwork calls for free...you basically route all your calls for free over gsm for a very low monthly price..??? |
18:23.16 | DyOS | am i on track or still confused? |
18:24.30 | marks_ | nope |
18:24.32 | lilneon | Dyos: no u getting it.. nice idea though.. so instead of iax between asterisk servers... use this to route your calls between them for free.. and ease up your broadband badnwidth as well for other things.. man u are brilliant!! |
18:24.33 | marks_ | it can |
18:25.36 | IQ | lilneon, DyOS, one active call per service provider at a time ? |
18:25.45 | DyOS | i would assume |
18:26.49 | IQ | even then can same some. Its good for called person too - at times they get free in network calls |
18:26.55 | G0shen | DyOS: 3000 minutes limit on most mobile to mobile plans |
18:27.03 | G0shen | or somewhere around there |
18:27.10 | DyOS | at&t was offering unlimited for awhile |
18:27.15 | IQ | G0shen: most in USA are offering unlimited |
18:27.18 | G0shen | yea "unlimited" |
18:27.28 | DyOS | but you could add unlimited nights and weekends and route calls during a certain time to that provider |
18:27.29 | G0shen | same as packet8's "unlimited" |
18:27.50 | G0shen | read the fine print, my AT&T contract has a limit on "unlmited" minutes |
18:28.01 | IQ | yeah - for non gsm as well :) |
18:28.20 | IQ | G0shen: talk to Cingular - they own AT&T now... Or switch to sprint pcs |
18:28.40 | DyOS | so how would it be of benefit to interface with gsm when i could get unlimited plan through voicepulse and call everywhere for free anyways...i don't see why you would want to route cellular |
18:28.45 | file[laptop] | we couldn't override their override of our override! |
18:28.55 | marks_ | CAN ANYONE HELP ME WITH ASTERISK INSTALL AND FWD ON GENTOO? PM ME, I WILL PAY A LITTLE!!! |
18:28.59 | marks_ | †|
18:29.09 | G0shen | no way, I get 7pm starting nights with my AT&T plan :) |
18:29.10 | file[laptop] | marks_: don't do that. |
18:29.15 | DyOS | hey marks you have a dedicated computer for *? |
18:29.17 | file[laptop] | marks_: there's guides out there, Google, |
18:29.20 | file[laptop] | just don't do that.. |
18:29.24 | marks_ | ive triied, ive tried |
18:29.35 | _Vile | heh |
18:29.42 | _Vile | it'd take me 20 seconds to find a url for you |
18:29.44 | _Vile | you haven't |
18:30.01 | marks_ | i need a step by step |
18:30.15 | IQ | marks_: Asterisk@Home |
18:30.16 | blitzrage | you don't get step by steps with Asterisk |
18:30.39 | IQ | marks_: just insert the CD and it'll do everything for u |
18:30.43 | marks_ | yea |
18:30.49 | marks_ | this is on a VSERVER at the planet |
18:31.13 | IQ | http://asteriskathome.sourceforge.net/ |
18:32.15 | IQ | is there any way we can receive SMS on our DIDs ? |
18:32.24 | file[laptop] | there's guides out there... just read the guides |
18:32.43 | IQ | file[laptop]: to receive SMS on DID ? |
18:33.06 | file[laptop] | IQ: you can't really... |
18:33.32 | *** join/#asterisk dudes (~dudes@12-215-34-6.client.mchsi.com) |
18:35.22 | *** join/#asterisk linagee (~linagee@netblock-66-245-227-138.dslextreme.com) |
18:35.31 | DyOS | marks you have a vserver at theplanet.com |
18:35.51 | marks_ | indeed... |
18:36.04 | linagee | ack. i can't get inbound sip calls working. :( my grandstream phone can do it all correctly. that is, when i punch the right user/pass and such into the grandstream i can receive calls, but i must be doing something wrong with asterisk |
18:37.08 | DyOS | what os is installed on it |
18:37.24 | linagee | DyOS: centOS. this is with asterisk@home. gah. |
18:37.25 | linagee | :( |
18:38.37 | linagee | when i call the number with sip debug on, i do see stuff scroll by, but it's not picking it up. |
18:39.21 | Zeeek | linagee the grandstream is registered with the asterisk box? |
18:39.30 | file[laptop] | marks_: you should just hire someone to set it up... because if you can't follow the guides out there, you're lost |
18:39.33 | linagee | Zeeek: yes. now it is. |
18:39.36 | marks_ | i kno |
18:39.39 | marks_ | how muc |
18:39.39 | marks_ | h |
18:39.43 | linagee | Zeeek: i think i need nat traversal on |
18:39.44 | marks_ | we are a non-profit |
18:39.56 | marks_ | and will give them credit on the pbx and an ad on each page of the site |
18:39.58 | Zeeek | linagee so what happens when you dial a number? What does the CLI say (with NO debug on) |
18:40.24 | linagee | Zeeek: when i dial out? it works perfectly. it has the caller id i put in and everything! :-) |
18:40.26 | Zeeek | linagee if you are traversing NAT yes |
18:40.39 | Zeeek | where can you not receivecalls FROM ? |
18:40.43 | linagee | Zeeek: i guess asterisk@home does not have that on by default |
18:40.52 | file[laptop] | marks_: find someone who will do it under the terms... there's a list at voip-info.org of consultants |
18:41.07 | linagee | Zeeek: i can't get inbound calls. my grandstream does not ring, my upstream provider voicemail kicks in. |
18:41.25 | Zeeek | where is the GS phone, where is asterisk ? |
18:41.32 | Zeeek | and who is calling you? |
18:41.39 | linagee | Zeeek: they are both behind a nat. my cell is calling me |
18:41.49 | bkw_ | yo yo yo |
18:41.51 | Zeeek | so the GS can dial |
18:41.54 | linagee | yes |
18:41.55 | _Vile | bkw YO |
18:41.56 | Zeeek | like the echo test |
18:42.07 | blitzrage | bkw_: yo hoe |
18:42.09 | linagee | Zeeek: and it registers apparently because i get sip debug junk when i ring in. |
18:42.17 | Zeeek | linagee your cell is calling you how? |
18:42.18 | Qwell | bkw_: You're just in time. file was about to take his muffins out of the oven. |
18:42.26 | linagee | Zeeek: my finger is pushing the numbers. :) |
18:42.47 | *** join/#asterisk mitmit (~mitmit@CPE000c418b4787-CM001225401fee.cpe.net.cable.rogers.com) |
18:42.57 | Zeeek | and after the nulbers, presumably there is a network connecting somhow to asterisk? How? |
18:43.13 | linagee | Zeeek: my voip provider does the pstn to voip |
18:43.25 | linagee | Zeeek: i am calling the number assigned to me |
18:43.48 | Zeeek | so it sounds like you havea problem at the incoming context you had @home set up for the voip provider |
18:43.51 | linagee | Zeeek: as i said, this is all fairly irrelavent as when my grandstream is setup, it does ring in. |
18:44.09 | file[laptop] | my muffins are hot and big |
18:44.13 | Zeeek | no it's not irrelevant |
18:44.19 | file[laptop] | would anyone like to butter them? |
18:44.21 | linagee | Zeeek: yes. this is true. my config here is probably foobar |
18:44.35 | Zeeek | what is irrelevant is trying to understand what's going on if you choose to use a precooked setup |
18:45.03 | Zeeek | usually you would know there is an area in the sip.conf where that voip thing is set up |
18:45.07 | linagee | Zeeek: i know a bit about asterisk |
18:45.13 | Zeeek | and you'd go there and see what is wrong with it |
18:45.17 | linagee | Zeeek: i just like asterisk@home's interface |
18:45.23 | Shido6 | AMP |
18:45.33 | Zeeek | sip show peers shows the voip prov? |
18:45.37 | linagee | Shido6: and that cool SWF thingy? :) |
18:45.39 | Shido6 | I should just fix AMP |
18:46.29 | Zeeek | linagee ok, since you know a little about asterisk you can post the relevant sections of the conf stuff for us |
18:46.33 | ariel_ | Shido6, there is an #amportal on here that can help to fix what you need in amp. There working on the new release right now. |
18:46.38 | Zeeek | and the CLI messages |
18:46.46 | linagee | Zeeek: Zeeek ok, it shows this |
18:46.48 | linagee | voipprovider_out/username voip.ip.addr 255.255.255.255 5060 Unmonitored |
18:46.48 | linagee | 200/200 192.168.2.108 D 255.255.255.255 5060 Unmonitored |
18:47.04 | linagee | !!! now you know my internal ip. shoot. lol |
18:47.13 | Zeeek | who is 200? |
18:47.16 | Zeeek | your GS? |
18:47.17 | linagee | my GS |
18:47.29 | Zeeek | so your peer entry for voip provider isn't set up |
18:47.35 | Aze` | Anyone have experience with HFC-S PCI Billion ISDN Card ? |
18:47.50 | Zeeek | do you register with that provider? - I guess so if the call comes in |
18:47.51 | linagee | Zeeek: should i have voipprovider_in? |
18:48.16 | Zeeek | how does the voip provider find you? With 192.168..... ? :) |
18:48.21 | linagee | Zeeek: well i don't know how far it comes in. i know that when i call i get voip junk |
18:48.37 | linagee | Zeeek: i have my NAT routing port 5060 TCP/UDP to the asterisk@home box |
18:48.41 | linagee | needing more ports? |
18:48.47 | Zeeek | maybe reading the junk would say something like "ALERT: Intruder trying to break in!" |
18:48.54 | linagee | lol |
18:49.11 | Zeeek | how do you expect the unknow provider to reach you? |
18:49.28 | Zeeek | what is there in sip.conf (it is SIP?) to identify it |
18:49.36 | linagee | second |
18:49.43 | Zeeek | who is this mysterious provider? |
18:49.46 | Zeeek | FWD? |
18:49.54 | linagee | it. hehege |
18:50.12 | linagee | Zeeek: a friend of mine. :) |
18:50.26 | linagee | Zeeek: he has too much money laying around or something so he has his own PRI. |
18:50.28 | linagee | lol |
18:50.39 | Zeeek | well, whatever *it* is, it needs to be identified by asterisk |
18:50.48 | Zeeek | Starter tutorial: |
18:50.48 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html |
18:50.48 | Zeeek | http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html |
18:50.48 | Zeeek | http://www.automated.it/guidetoasterisk.htm |
18:50.48 | Zeeek | THE reference of the moment: |
18:50.48 | Zeeek | http://www.asteriskdocs.org |
18:51.29 | linagee | ok. it puts everything into sip_additional.conf |
18:53.25 | linagee | ok |
18:53.27 | Zeeek | linagee I tried to help you. You answer questions with jokes. Do some reading and come back when you can express the problem in terms someone can understand |
18:53.33 | linagee | i have a cooked config that it generated |
18:53.49 | linagee | aha! |
18:53.52 | linagee | nat=never? wtf??? |
18:54.19 | Shido6 | . |
18:54.24 | Zeeek | .. |
18:54.30 | Qwell | ,. |
18:54.31 | Zeeek | .... |
18:54.34 | Shido6 | ! |
18:54.36 | Zeeek | >>>>>>>>>>>>>>>>>> |
18:54.52 | *** join/#asterisk marlowe (~marlowe@bmw.princetonhost.com) |
18:56.07 | Aze` | Anyone have experience with HFC-S PCI Billion ISDN Card, pls? |
19:01.50 | Blackvel | what is the problem? i have a hfc (different). should not really matter with zaphfc |
19:02.54 | *** join/#asterisk IQ (~iq@70-59-165-40.omah.qwest.net) |
19:04.12 | Aze` | Blackvel > in PTP mode, where try to dial Dial(ZAP/g1/xxxxxxx), it's open channel and i listen tone but it dont compose number.. why ? |
19:04.21 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
19:04.58 | Aze` | Blackvel > same problem with HiSax |
19:05.19 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.rr.com) |
19:05.26 | IQ | anyone using Festival with * ? |
19:05.37 | marks_ | FCUK |
19:05.40 | marks_ | FUCK |
19:05.54 | niZon | IQ: I tried... |
19:05.58 | niZon | and failed |
19:06.07 | marks_ | FUCK |
19:06.10 | marks_ | ok |
19:06.10 | drumkilla | I use it - but it soudns terrible :) |
19:06.13 | *** part/#asterisk marks_ (~Marks_Dis@cpe-70-112-81-84.austin.res.rr.com) |
19:06.15 | IQ | niZon: so maybe my turn to try :P |
19:06.22 | IQ | drumkilla: what version do u use? |
19:06.30 | niZon | yeah |
19:07.36 | drumkilla | heck, i don't know ... |
19:07.42 | IQ | oh :) |
19:07.44 | drumkilla | but check out Cepstral |
19:07.48 | drumkilla | it's only $30 |
19:07.55 | drumkilla | but sounds a loooot better |
19:08.02 | IQ | the current version that is available on their website is for Free and is a BETA version |
19:08.06 | niZon | is it easy to get running? |
19:08.18 | *** join/#asterisk fixitjimmy (~aficionar@dsl82-163-227-225.as15444.net) |
19:08.24 | niZon | (directed towards drumkilla) |
19:08.31 | drumkilla | from what I hear, yeah |
19:08.36 | drumkilla | I haven't paid for it yet, heh |
19:08.37 | Aze` | Blackvel, are u there ? |
19:08.40 | Blackvel | aze: hisax? zap and hisax? uh? |
19:08.42 | niZon | ah |
19:10.11 | IQ | if Cepstral is better than Festival then :(( |
19:10.53 | Zeeek | I have Cepstral |
19:11.16 | Zeeek | it is easy to get it working - and it sounds *decent* but not great |
19:11.23 | IQ | I use ScanSoft's Speechify - it sounds better than I speak :P |
19:11.43 | Zeeek | I have only used it with pre-digested text files already made into wav |
19:11.47 | niZon | hm, is there a guide for getting it working with * |
19:11.48 | niZon | ? |
19:11.56 | niZon | oh |
19:12.20 | IQ | Zeeek: speechify or cepstral? |
19:12.40 | Zeeek | Cepestral - speechify I couldn't find at the time |
19:12.46 | Zeeek | it was bought out by someone |
19:13.24 | Zeeek | is there a speechify product to be used with asterisk? |
19:13.55 | *** join/#asterisk IQ (~iq@70-59-165-40.omah.qwest.net) |
19:14.12 | *** join/#asterisk sgaiadog (~josh@24-29-142-30.nyc.rr.com) |
19:19.28 | IQ | ScanSoft's online demo: http://www.scansoft.com/speechworks/realspeak/demo/default.asp |
19:19.56 | Zeeek | do they sell a simple product to be useable with *? I never found one but the voice is superior |
19:20.12 | Zeeek | I remember having say "dirty" things :) |
19:20.37 | Zeeek | Find the Valley Girl and ask her to ask you anything! |
19:21.00 | IQ | we use SpeechWorkd's Speechify at work. |
19:21.30 | IQ | I wrote a DLL to interface with their engine. but nothing so far for * |
19:22.08 | niZon | do it with AGI |
19:22.18 | niZon | make it create a wave then use playback perhaps? |
19:22.21 | *** part/#asterisk sgaiadog (~josh@24-29-142-30.nyc.rr.com) |
19:22.33 | niZon | depending on how long it takes to generate the wav |
19:22.39 | *** join/#asterisk sgaiadog (~josh@24-29-142-30.nyc.rr.com) |
19:22.52 | Zeeek | Go to Jill and type in "Hey, I saw one once that was a lot smaller" |
19:22.54 | IQ | niZon: this is not bad - we use this method in few apps that can't call dlls |
19:23.06 | niZon | hm |
19:23.38 | linagee | Zeeek: damn it. i hate it when you're right. :) "SIP/2.0 403 Forbidden" |
19:23.52 | linagee | Zeeek: that was in all the "junk" that came in on an incoming call. |
19:24.05 | IQ | loool |
19:24.20 | IQ | they have Indian English :O |
19:24.22 | Zeeek | IQ the speechify voices sound great but I've never found a linux product to play them |
19:25.00 | IQ | Zeeek: they do support Linux - call them :) |
19:25.07 | Zeeek | but not for $30 ? |
19:25.14 | IQ | Zeeek: no way |
19:25.34 | Zeeek | the "call us" made me think "starting at $500" or worse |
19:25.40 | Zeeek | the quality is excellent |
19:25.41 | IQ | Zeeek: once I asked them for a development kit - the dude said that it will cost me 1600 to get dev kit |
19:25.49 | Zeeek | that's more like it |
19:25.59 | IQ | Zeeek: then they bought it at work- so I use it for free :) |
19:26.11 | Zeeek | maybe we can all pitch in $32 (50 of us)? |
19:26.26 | IQ | Zeeek: Try "English, Indian - Sangeeta" she sounds a lot better |
19:26.38 | Zeeek | well the demo allows you to put together a whiole set of great sounding files for asterisk |
19:27.00 | IQ | Zeeek: yeah - but there is a limit on length of sentense |
19:27.01 | Zeeek | Maybe Dell support should lok into that one ;) |
19:27.21 | IQ | max 100 characters |
19:27.35 | Zeeek | All my sex phrases were shorter than that |
19:27.41 | IQ | loool |
19:27.42 | Zeeek | like "blow me" |
19:27.52 | IQ | But youc an always generate multiple phrases and then join them |
19:27.56 | IQ | *you can |
19:28.07 | Zeeek | "Oooooooh ohhhhh yes, yes, yesy YES!" |
19:28.16 | Qwell | yesy? |
19:28.35 | Zeeek | yeah well, TTS breaks down sometimes on emotional stuff |
19:29.38 | *** join/#asterisk b0ef (~b0ef@062016141085.customer.alfanett.no) |
19:29.39 | IQ | no sample .wav for Festival ? |
19:29.55 | Zeeek | Never tried ti |
19:29.57 | Zeeek | it |
19:30.47 | b0ef | is it possible to change speex encoding parameters during a call from the asterisk cli? |
19:31.04 | Zeeek | I had Cepestral's Dave read a tex from a food magazine. "Nothing is harder to make than good fried chicken thant's not greasy." |
19:31.26 | Zeeek | After the first sentence, it really sounded stupid |
19:31.32 | linagee | someone write a plugin to steal TTS off that one AT&T website. that thing makes very realistic .wav files! :) |
19:31.50 | IQ | linagee: link please |
19:32.08 | b0ef | linagee: not as long as it's not free;) |
19:32.19 | linagee | IQ: google. http://www.research.att.com/projects/tts/demo.html |
19:32.57 | linagee | b0ef: just make the module say something like, "you understand and agree to all terms <at the above url> press 'y' to proceed" lol. :) |
19:33.01 | Zeeek | that's not as good as the other |
19:33.07 | linagee | Zeeek: where is the other |
19:33.22 | Zeeek | <PROTECTED> |
19:33.38 | IQ | linagee: ATT sound good :) |
19:33.50 | linagee | Zeeek: does the other one generate .wav files? :) |
19:33.54 | IQ | and no 100 char limit either |
19:34.06 | Zeeek | I think they're way, but I don't remember |
19:34.09 | linagee | oh. i guess it does. |
19:35.14 | Zeeek | The ATT has better French though |
19:35.22 | IQ | distortion in ATT wav |
19:36.17 | *** join/#asterisk KingMorse (~nowhere@201-108-141-208.tranquility.net) |
19:36.27 | KingMorse | hi everyone |
19:37.44 | KingMorse | anyone around that can help me troubleshoot a CallerID issue? |
19:38.05 | Zeeek | go for it KingMorse |
19:39.14 | KingMorse | well I'm getting some errors when receiving calls |
19:39.21 | KingMorse | a series of 3 actually |
19:39.33 | KingMorse | I don't know how to use pastebin so I'll just describe them here |
19:39.41 | KingMorse | one says "fsk_serie made mylen..." |
19:39.48 | KingMorse | the next is "CallerID feed failed" |
19:39.57 | Zeeek | KingMorse what country are you in? |
19:40.02 | KingMorse | then "CallerID returned with error on channel 'Zap/3-1' |
19:40.08 | KingMorse | I'm in the United States |
19:40.10 | Nugget | "<KingMorse> I don't know how to use pastebin so..." <-- the point where I lost interest in helping. |
19:40.22 | Zeeek | try adding a wait(2) and see if that makes any diff |
19:40.27 | KingMorse | I'm sorry Nugget I heard of pastebin and i know it's polite to use but I don't know how to use it |
19:40.33 | KingMorse | forgive me for not being an IRC master |
19:40.45 | Nugget | if you're too lazy to learn how to paste then there's no hope for you to ever get asterisk working. |
19:40.46 | Qwell | ~pastebin |
19:40.47 | jbot | well, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
19:40.50 | Zeeek | just go there and paste your stuff but first try adding a wait(2) |
19:41.03 | Nugget | because asterisk is a lot harder to use than your paste button. |
19:41.03 | KingMorse | a wait(2), will try that hold on a sec |
19:41.05 | Zeeek | before the rest of the extension |
19:41.10 | Zeeek | of even 3 |
19:41.15 | Zeeek | or |
19:41.25 | KingMorse | hey Nugget, I have Asterisk running just fine thankyouverymuch |
19:41.35 | Nugget | you don't have to be an irc master, but you do have to show some initiative that you actually care about solving your problem. |
19:41.38 | KingMorse | but I prefered to get to the point rather than ask about pastebin first |
19:41.50 | Nugget | if you don't care, why should we? |
19:42.21 | KingMorse | actually Nugget, I'm putting you on ignore...have a nice day |
19:42.33 | Nugget | good luck finding people to do you job for you. |
19:42.40 | linagee | Nugget: if you were an irc master you would not need an irc client. just talk to the servers through telnet. you could do DCC transfers through telnet too. :) |
19:42.41 | Qwell | Nugget: I think he just proved your point for you. |
19:42.49 | Zeeek | oh, Nugget... you're in the shit now.... |
19:42.59 | Zeeek | put the shields up |
19:43.15 | Nugget | linagee: yeah, but I get tired of typing PONG all the time. :) |
19:43.20 | linagee | hehehe |
19:43.44 | KingMorse | hm, zeeek looks like the same issue |
19:43.56 | Zeeek | I've seen that error myself a few times |
19:44.13 | Zeeek | it's a frank message : "wtf? I don't get that cid that was sent at all..." |
19:44.31 | KingMorse | well I've confirmed that CallerID is set up on the line |
19:44.38 | KingMorse | it works with a regular phone |
19:44.43 | Zeeek | what are you looking at? X100P ? clone? |
19:44.55 | KingMorse | i'm running a wildcard TDM400 |
19:45.02 | KingMorse | with 3 fxo modules |
19:45.15 | *** join/#asterisk J[SS] (~jeremy@chaoscon.user) |
19:45.31 | Zeeek | and obviously usecallerid and all that in zapata |
19:45.40 | KingMorse | correct |
19:45.58 | Zeeek | what version of asterisk? |
19:46.18 | KingMorse | tried several, currently the most recent CVS-HEAD |
19:46.34 | Zeeek | same error in like 1.0.5 or 6? |
19:46.39 | Zeeek | STABLE? |
19:46.56 | KingMorse | yes |
19:47.49 | Zeeek | and the FXO work fine other than this issue? |
19:47.54 | Zeeek | no noise ? |
19:48.19 | KingMorse | yes although I have had issues with echo, they have been largely resolved through the use of fxotune and the echocancel config settings |
19:48.26 | *** join/#asterisk Sedorox (~Sed@Neptune-W.client.wlgrv.pa.sed6.net) |
19:49.01 | Zeeek | I'm trying to remember when I saw those errors |
19:49.15 | *** join/#asterisk ckruetze (~nospam@i3ED63E95.versanet.de) |
19:50.56 | Zeeek | KingMorse look this up: cidsignalling |
19:52.16 | KingMorse | hm, you mean on google? |
19:52.25 | CoaxD | Man, learning ASL is tough |
19:52.27 | Zeeek | whatever |
19:52.47 | blitzrage | Adult Sign Language? |
19:52.47 | *** join/#asterisk Jer13261 (~Jer@rdu57-251-152.nc.rr.com) |
19:52.48 | CoaxD | blitzrage: That'd be "American", but yea |
19:52.54 | blitzrage | errrr... american :) |
19:52.59 | Zeeek | Adult is more fun though |
19:52.59 | KingMorse | hm, what should cidsignalling be for US? |
19:53.13 | CoaxD | blitzrage: I'm learning how to fingerspell right now |
19:53.21 | blitzrage | CoaxD: yah.. that's hard to get good at :) |
19:53.23 | Zeeek | KingMorse bell? I don't knwo, but there are three possibilities I think |
19:53.25 | CoaxD | blitzrage: First things first.. you know. (Gotta be able to spell to make words) |
19:53.25 | blitzrage | CoaxD: you drive? |
19:53.31 | CoaxD | blitzrage: I do drive |
19:53.59 | blitzrage | CoaxD: when driving, good practise is spelling all the street names |
19:53.59 | blitzrage | you'll get good fast. |
19:53.59 | CoaxD | blitzrage: Oh???? |
19:53.59 | CoaxD | blitzrage: THANK YOU!!! :) |
19:54.02 | blitzrage | CoaxD: np! |
19:54.04 | CoaxD | blitzrage: That is an EXCELLENT suggestion! |
19:54.13 | blitzrage | CoaxD: just spell any sign you look at. |
19:54.16 | KingMorse | i've done the same thing with morse code |
19:54.28 | CoaxD | blitzrage; (Plus, you're a geek, so chances are, our brains work just a wee bit more alike than i to other more non-technologically oriented people) |
19:54.38 | CoaxD | KingMorse; You know, perhaps thats why I could never learn morse either? |
19:54.38 | *** join/#asterisk D1ng0 (~dingo@26.37.33.65.cfl.res.rr.com) |
19:54.46 | blitzrage | CoaxD: I had to take some courses when I was teaching soccer to a deaf child |
19:54.50 | D1ng0 | anyone here living in the phillipiness ??? |
19:55.01 | CoaxD | blitzrage: Oh man! Wow! |
19:55.02 | Zeeek | KingMorse I don't have an answer but a start is to google the error messages. You'll see a lot of stuff from mailing list, and maybe you'll find the asnwer |
19:55.07 | CoaxD | blitzrage: Not too many people would do that, sir |
19:55.15 | KingMorse | yeah I've been through that already zeeek |
19:55.23 | KingMorse | I *do* care about solving my problem ;) |
19:55.27 | Zeeek | I wish I had written down what made error message go away each time :( |
19:55.35 | CoaxD | blitzrage: So apparently, aside from learning another language, you also have an immense appreciation for children. |
19:55.41 | blitzrage | CoaxD: yah. I coached soccer from when I was 16-22 |
19:55.47 | file[laptop] | blitzrage is O-L-D |
19:55.51 | blitzrage | CoaxD: this is true. But I don't want any :) |
19:55.53 | blitzrage | file[laptop]: lol |
19:56.00 | CoaxD | blitzrage: Hehe :) I have a 4 year old |
19:56.02 | Zeeek | blitz is way younger than me |
19:56.15 | CoaxD | except zeek |
19:56.21 | CoaxD | zeek is what, 60? |
19:56.25 | file[laptop] | FEAR ME! |
19:56.30 | Zeeek | Be afraid |
19:56.46 | file[laptop] | blitzrage is also VERY silly |
19:57.11 | blitzrage | CoaxD: I like kids... just not my own. |
19:57.43 | CoaxD | blitzrage: there is nothing wrong with that, sir |
19:59.23 | blitzrage | CoaxD: who knows, maybe some day. But with the way the world is going, its not going to be appropriate to bring a child into the world. |
19:59.35 | CoaxD | blitzrage: I concur entirely. |
19:59.45 | file[laptop] | doesn't mean you can't have fun in the mean time |
20:00.23 | blitzrage | file[laptop]: never said I wasn't having fun :) |
20:00.28 | file[laptop] | haha |
20:00.32 | *** join/#asterisk vidia (dgcddert45@ip68-12-230-54.ok.ok.cox.net) |
20:00.50 | blitzrage | file[laptop]: hell.. ask #asterisk-doc about that bed I broke once... I don't think I'll ever hear the end of that one |
20:00.51 | Qwell | ... |
20:01.18 | blitzrage | Qwell: its funny because we weren't doing anything "fun" |
20:01.28 | Qwell | mmhmm |
20:01.30 | file[laptop] | was it leading up to "fun" |
20:01.35 | zoa | hello there |
20:01.36 | Qwell | :p |
20:01.45 | file[laptop] | zoa! |
20:03.55 | blitzrage | file[laptop]: unfortunately not... |
20:04.06 | *** join/#asterisk flewid (~flewid@24.42.244.169) |
20:04.08 | flewid | sup all |
20:04.09 | blitzrage | actually... it might have... I forget... but she was great. |
20:04.09 | file[laptop] | blitzrage: poor you |
20:04.10 | blitzrage | lol |
20:04.12 | file[laptop] | lol |
20:04.16 | file[laptop] | she was "great" eh? |
20:04.28 | zoa | thats like she was nice |
20:04.29 | flewid | quick question - with the voicemail alerts that get emailed, i know i can define seperate contexts within voicemail.conf |
20:04.37 | flewid | but can those seperate contexts also have a seperate message? |
20:04.52 | flewid | (reason being we run a few companies off the same system, and they want the emails tailored to them) |
20:05.01 | *** join/#asterisk [hC] (~turnerd@8.10.2.4) |
20:05.25 | *** part/#asterisk b0ef (~b0ef@062016141085.customer.alfanett.no) |
20:05.53 | [hC] | If i neglect to specifically set CIDName and CallerID, will Dial() use whatever my phone passes along instead? |
20:06.11 | [hC] | or whatever is set in sip.conf, i suppose. |
20:06.31 | Jer13261 | wow * seg faults when i call it from my C7960 |
20:06.41 | Jer13261 | yes |
20:07.00 | flewid | Jer13261: interesting |
20:07.17 | [hC] | I suppose it goes in the priority of phone, sip.conf, then manual SetCIDName? |
20:07.35 | flewid | [hC]: whatever the last place it was set i believe |
20:07.36 | Mw3 | i have a serious problem. we want to deploy voip services on our network (about 1000 users on a housing estate) but the local authorites requires that it has to be unspoofable/sniffable. so i think we have 2 choises, change or switches to manageable and do portsecurity and etc... (expensive, we have only about 12 manageable switch on the core points) or find some vpn capable ata devices (cpe). is there anythink which could do vpn ? or do you have any other ... |
20:07.43 | Mw3 | ... ideal to make it right for the fucking authority ? :) |
20:07.50 | niZon | Jer13261: potential remote exploit? :\ |
20:08.16 | Jer13261 | yea ..... |
20:08.17 | flewid | Mw3: wait for iax2's encryption to work as it should? or yeah, a vpn |
20:08.23 | flewid | i think the sipuras support vpn's don't they? |
20:08.23 | Jer13261 | ? |
20:08.40 | Mw3 | flewid: i'm going to check |
20:08.41 | file[laptop] | iax2 encryption, THAT'S what I was going to muck with |
20:08.49 | file[laptop] | ...later |
20:08.55 | Nugget | no! do it now! :) |
20:08.57 | flewid | file[laptop]: i knew i had a purpose! |
20:08.58 | flewid | :) |
20:09.24 | file[laptop] | hehe |
20:09.24 | Jer13261 | how can i debug this err? |
20:09.32 | file[laptop] | okay fine people |
20:09.34 | flewid | what's the error |
20:09.40 | file[laptop] | lemme get a test environment setup |
20:09.44 | Jer13261 | * seg faults |
20:09.51 | Mw3 | flewid: by the way, whats wrong with iax2 encryption ? |
20:09.53 | Jer13261 | when called via sip |
20:10.02 | flewid | Jer13261: what are you doing, is debug on? did it work before? have you upgraded recently? |
20:10.19 | flewid | Mw3: i couldn't tell you, i haven't had much time to actually play with it yet |
20:10.20 | flewid | :( |
20:10.31 | flewid | been more involved to switching our systems to use realtime |
20:10.35 | flewid | well, our test systems |
20:10.39 | Jer13261 | Asterisk CVS-HEAD-03/26/05-07:04:19 |
20:10.48 | flewid | okay |
20:10.57 | flewid | so you're using head, have you tried reverting back to a version that worked? |
20:11.02 | Jer13261 | i thikn this could be a realtime issue since realtime doesnt work via odbc i had to use mysql :( |
20:11.03 | flewid | head isn't guaranteed to be stable in any way, nor even work |
20:11.24 | Jer13261 | true |
20:11.56 | flewid | %#> show version |
20:11.56 | flewid | Asterisk CVS-HEAD-03/26/05-03:10:04 built by root@asterisk on a i686 running Linux |
20:12.05 | flewid | i'm using that, and it's working with realtime |
20:12.14 | flewid | maybe just try checking out again |
20:12.20 | Jer13261 | alright |
20:13.12 | Jer13261 | doing that now :) |
20:13.16 | flewid | :) |
20:13.30 | flewid | also, go to logger.conf, and change console to include 'debug' |
20:13.38 | flewid | then asterisk -rx "stop now" |
20:13.45 | flewid | then asterisk -vvvvvvvvvvvvvvgc and check out the debug outpu |
20:15.03 | Jer13261 | compiling now |
20:15.32 | flewid | we finally got our portage server, now i get to see how installing gentoo on a raid1 system goes remotely |
20:15.33 | flewid | :) |
20:16.37 | KingMorse | hm, still no luck on the callerid front |
20:17.03 | KingMorse | does anyone know what the proper settings are for cidstart and cidsignalling in the US? |
20:17.14 | flewid | PRI? or ZAP? |
20:17.18 | KingMorse | zap |
20:17.24 | KingMorse | wildcard tdm400 |
20:17.29 | flewid | hmm, i'm in canada, and i didn't have to change anything |
20:17.29 | flewid | sec |
20:17.33 | *** join/#asterisk ikey (ikey@220.226.52.252) |
20:17.51 | flewid | usecallerid=yes |
20:17.51 | flewid | hidecallerid=no |
20:17.53 | flewid | from zapata |
20:18.15 | flewid | then callerid=asreceived |
20:18.20 | Jer13261 | ok now no crash :) |
20:18.28 | flewid | within the channel definition for my x100p card |
20:18.42 | flewid | and then i think i had to add a 2 second wait to the main menu so it would grab that callerid info |
20:18.56 | flewid | Jer13261: glad to hear it :) |
20:19.24 | KingMorse | unfortunately I'm getting these errors |
20:19.25 | KingMorse | http://www.mypastebin.com/?code=738791200 |
20:19.28 | KingMorse | any ideas flew? |
20:20.39 | flewid | sec |
20:20.55 | flewid | hmm |
20:21.14 | flewid | you have a callerid on that line right? :) (i know, but sometimes people forget the easy stuff) |
20:21.23 | KingMorse | yes, I do :) |
20:21.26 | flewid | and what version of * you using? HEAD ? |
20:21.28 | Jer13261 | can someone tell me what USE_POSTGRES_VM_INTERFACE= does? |
20:21.32 | KingMorse | yes, HEAD |
20:21.42 | flewid | Jer13261: i think that's the oldschool method of doing mysql-voicemail |
20:21.46 | flewid | i didn't have to change that with realtime |
20:21.55 | flewid | i could be wrong tho - it might be for the web interface |
20:22.02 | Jer13261 | ok i need the filename of the voicemail in a db someplace |
20:22.07 | Jer13261 | can that be done? |
20:22.15 | flewid | yeah use the ast_config table |
20:22.19 | ikey | does any one know how many calls per second can asterisk handle in realtime |
20:22.25 | flewid | and the add config files scripts that bkw made |
20:22.27 | flewid | i think it was bkw |
20:22.34 | ikey | on sip/h323/pstn |
20:22.38 | flewid | ikey: depends on servers i believe |
20:22.46 | flewid | and redundancy, and codecs, and etc |
20:22.59 | flewid | KingMorse: hmm, they just changed the callerid stuff to have that cdr_custom.so |
20:23.03 | flewid | perhaps it's something ot do with that? |
20:23.10 | flewid | i put noload cdr_custom.so in my modules.conf |
20:23.15 | flewid | cause it wasn't working for me |
20:23.17 | ikey | say dual processor server with intel xeon + 2gb ram on failover? |
20:23.25 | ikey | and G.711 codec |
20:23.32 | *** join/#asterisk anachron (~phr34k@ip70-176-146-245.ph.ph.cox.net) |
20:23.34 | flewid | ikey: i think on voip-info they have 350 simultaneous calls on a server like that |
20:23.41 | flewid | but it depends on everyone's situation |
20:23.57 | zoa | dont trust voip info for how many calls you can do |
20:24.02 | flewid | lol |
20:24.11 | flewid | don't trust anyone |
20:24.14 | flewid | trust yourself and your testing |
20:24.23 | Qwell | Don't even trust yourself. |
20:24.38 | flewid | well, hehe, trust yourself, but add leeway |
20:24.39 | ikey | :) |
20:25.08 | ikey | will it support atleast 200? |
20:25.11 | flewid | KingMorse: did you try adding debugging in? |
20:25.19 | flewid | ikey: it should on a server like that |
20:25.22 | flewid | that's pretty beefy |
20:25.29 | Jer13261 | that allows to store the users in a table i want to store messages |
20:25.39 | flewid | Jer13261: voicemail messages in db? |
20:25.45 | Jer13261 | yeppp |
20:25.52 | flewid | not gonna happen afaik |
20:25.58 | Jer13261 | well not the file itself but a record of the file |
20:26.01 | Qwell | Shouldn't be hard to code up |
20:26.06 | flewid | oh |
20:26.16 | flewid | yeah you should be able to add something in there as to which file was written |
20:26.19 | flewid | don't think it's possible right now tho |
20:27.12 | KingMorse | flewid: what do you mean add debugging in? at compile time? |
20:27.15 | Jer13261 | ok well my problem is i cant work out the filename of the voice mail that got left for a user heh |
20:27.38 | flewid | KingMorse: in logger.conf, toss 'debug' behind the 'console' line |
20:27.43 | flewid | so you can see the debug output as you run it |
20:29.40 | cjk | hi, lets say I do a dial sip/username1&sip/username2 . username1 is on the phone (busy) username2 is unavailable. what will be the value of dialstatus?a |
20:31.05 | KingMorse | hm |
20:31.16 | KingMorse | ok well nothing looks useful |
20:31.27 | KingMorse | also, the error messages in that pastebin aren't happening |
20:31.35 | KingMorse | they come and go |
20:31.44 | KingMorse | but at no time does CID seem to want to work |
20:33.20 | linagee | asterisk inside NAT, clients inside NAT is hard? |
20:33.32 | linagee | (two different NATs) |
20:34.32 | niZon | sounds painful |
20:36.38 | cjk | is there a way to disable in voicemailmain the busy and temporary message options. i want to disable them |
20:38.59 | Moc | everything is possible |
20:39.17 | drumkilla | it wouldn't be easy ... |
20:39.20 | drumkilla | i'm not sure why that would be worth it |
20:39.41 | BuckRogers | hey has anyone incontered a problem with the grandstream Handytone 486 with ip registration, it will not take a static or dynamic ip scheme |
20:39.47 | flewid | drumkilla: is it possible to have the email voicemail alerts be different for each voicemail context? |
20:39.58 | flewid | like [default] has email1, [companya] has email2 |
20:39.59 | flewid | ? |
20:40.05 | drumkilla | ooh ... good question |
20:40.13 | drumkilla | I don't think so, but that would be a nice feature |
20:40.33 | zoa | drumkilla: digium support doesnt reply to my email :( |
20:40.48 | BuckRogers | just call digium support |
20:40.52 | flewid | :) yeah we have a client asking for his emails to be different than the ones we're sending out currently |
20:40.52 | BuckRogers | their great guys |
20:40.55 | flewid | i wasn't sure if that was possible or not |
20:40.59 | flewid | guess now i know :) |
20:41.00 | flewid | thanks man |
20:41.08 | drumkilla | np |
20:41.21 | BuckRogers | hey has anyone incontered a problem with the grandstream Handytone 486 with ip registration, it will not take a static or dynamic ip scheme |
20:41.27 | zoa | i dont want to call them :) |
20:41.33 | zoa | i want an email back :( |
20:41.39 | drumkilla | they will eventually |
20:41.40 | drumkilla | it's the weekend |
20:41.42 | BuckRogers | why not they have helped our technical staff a load |
20:41.43 | flewid | you can't always get what you want! :) |
20:41.46 | flewid | and it's easter right now |
20:41.48 | zoa | i sent it 3 days ago or so |
20:41.50 | zoa | :) |
20:41.55 | drumkilla | hm .. :( |
20:42.00 | BuckRogers | dont for get that it is a holiday weekend for some |
20:42.05 | zoa | its just some hard questions on te410ps etc :) |
20:42.07 | drumkilla | i'm sure they'll get back to you early this week |
20:42.14 | BuckRogers | such as? |
20:42.29 | zoa | t1e1override |
20:42.38 | zoa | lspci output for all the digium cards |
20:42.41 | zoa | such things :) |
20:43.18 | BuckRogers | zoa: t1 e1 overfide is the jumper setting? |
20:43.21 | zoa | firmware numbers for the cards |
20:43.23 | BuckRogers | i beleive |
20:43.29 | zoa | its an override for the jumper setting |
20:43.33 | zoa | i think i figured it out |
20:43.41 | zoa | that it accepts a digit between 0 and 15 |
20:43.45 | zoa | and those are bits |
20:43.48 | BuckRogers | a software over ride for the hardware |
20:43.52 | zoa | where 0 = t1 and 1 = e1 |
20:43.56 | zoa | just want a confirmation |
20:44.16 | BuckRogers | in the zaptel drivers..? |
20:44.21 | zoa | nah |
20:44.25 | zoa | when loading the module |
20:44.27 | BuckRogers | where about |
20:44.33 | zoa | its not in there |
20:44.45 | zoa | and i think the firmware == the FALC version |
20:44.46 | zoa | but unsure |
20:45.07 | BuckRogers | you want to pass in parematers while loading the zap moduel? |
20:45.22 | zoa | when loading the driver module |
20:45.36 | zoa | i can use it |
20:45.39 | BuckRogers | is that apzap? |
20:45.40 | zoa | just want confirmation |
20:45.50 | zoa | that would be like wct4xxp |
20:46.06 | Blackvel | hey |
20:46.13 | BuckRogers | literally the hardware specific drivers |
20:46.17 | zoa | im also documenting all errors you might get with asterisk |
20:46.19 | zoa | and the solution |
20:46.31 | Blackvel | I get always this reply from AGI (I didn't send any specific command like this): 510 Invalid or unknown command |
20:46.35 | Blackvel | what's wrong? |
20:46.36 | BuckRogers | sounds like an intresting post |
20:46.56 | zoa | it will take some more months |
20:47.14 | BuckRogers | i could imagine |
20:47.48 | Mw3 | another problem. is there any signalling gateway which do ss7 <-> sip ? |
20:48.21 | BuckRogers | yeah asterisk |
20:49.03 | Mw3 | hm, i've read that asterisk-ss7 just in progress |
20:49.11 | BuckRogers | hey has anyone incontered a problem with the grandstream Handytone 486 with ip registration, it will not take a static or dynamic ip scheme |
20:49.17 | BuckRogers | so what is the problem |
20:49.30 | zoa | asterisk-ss7 is there |
20:49.30 | BuckRogers | we do it now |
20:49.35 | zoa | but the cost is unknown |
20:49.53 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
20:50.08 | BuckRogers | ss7 singnaling to and from sip |
20:50.13 | BuckRogers | ..? |
20:50.22 | BuckRogers | its the purpose of diguim |
20:50.26 | zoa | to and from anything |
20:50.27 | BuckRogers | hardware |
20:50.31 | zoa | its not digium stuff |
20:50.45 | BuckRogers | yeah but they support it |
20:50.48 | BuckRogers | so does cisco |
20:50.50 | zoa | i dont think they do |
20:51.10 | BuckRogers | i dont think i understand what he is saying, could you re phrase it |
20:51.34 | zoa | i mean that the ss7 for asterisk implementation is not done by digium |
20:51.38 | zoa | and not sold by digium |
20:51.42 | zoa | but by a 3rd party |
20:51.50 | zoa | but you will need a license for asterisk |
20:51.54 | BuckRogers | whom is this 3rd party |
20:51.55 | zoa | if you want to buy the ss7 |
20:52.00 | zoa | a german guy |
20:52.04 | zoa | marku something |
20:52.35 | BuckRogers | so what happens when you buy a te405p and have a data and a pri t1 |
20:52.48 | BuckRogers | and want to go from ss7 to sip |
20:53.26 | zoa | you need to buy a license from digium |
20:53.31 | zoa | and one from marku |
20:53.38 | BuckRogers | is it included in the price of the card? |
20:54.01 | BuckRogers | im assuming? |
20:54.09 | zoa | no no no no |
20:54.10 | *** join/#asterisk dudes (~dudes@12-215-34-6.client.mchsi.com) |
20:54.18 | zoa | it has nothing to do with the card |
20:54.46 | dudes | Anyone know why asterisk CLI programs may not work? hangup,dial,softhangup, ect. |
20:54.48 | BuckRogers | really so how does a company like voicepulse that uses diguim hardware and sip with pri functino they have bought the lincense |
20:55.06 | zoa | you dont need ss7 to use a pri card |
20:55.12 | zoa | pri also exists without ss7 |
20:55.14 | BuckRogers | ahhh i see |
20:55.21 | zoa | pri cards also do t1 or e1 |
20:55.36 | BuckRogers | t1 carry ss7 signaling |
20:55.59 | BuckRogers | North american standard |
20:56.21 | zoa | no they dont |
20:56.33 | zoa | they can carry ss7 |
20:56.35 | zoa | but they dont need to |
20:57.39 | BuckRogers | yeah thier is common channel and out of band also |
20:58.05 | BuckRogers | but ss7 is old school still in use |
20:58.45 | BuckRogers | anyhow has any one hooked up a grandstream 486 here |
20:59.15 | Blackvel | who can help me with AGI? |
20:59.16 | Blackvel | <message>Sending [192.168.1.2] : EXEC DIAL ZAP/g1/90|30|tr |
20:59.22 | Blackvel | when I send this |
20:59.32 | Blackvel | <message>Got BINARY [192.168.1.2] : 200 result=0 |
20:59.34 | Blackvel | I get this reply |
20:59.44 | Blackvel | which indicates that everything is okay (my zap telephone rings) |
20:59.56 | Blackvel | but when I read the input stream, another messages appears |
21:00.07 | Blackvel | <message>Got BINARY [192.168.1.2] : 510 Invalid or unknown command |
21:00.30 | Blackvel | but I don't send any other commands after dial, so, why do I receive a 510 invalid? |
21:01.46 | Blackvel | do I get another reply when I hangup the telephone? |
21:02.02 | Blackvel | like when the dail returns? |
21:02.12 | Blackvel | I receive this at the very end after the pickup |
21:02.16 | Blackvel | err hangup |
21:02.18 | Blackvel | <message>Got BINARY [192.168.1.2] : 200 result=-1 |
21:04.46 | Shido6 | zZzz |
21:05.00 | Blackvel | yes |
21:05.01 | Blackvel | :) |
21:05.09 | Blackvel | do you feel like this? |
21:05.20 | *** join/#asterisk tainted- (~ta_i_nted@65-60-70-243-cust.telepacific.net) |
21:08.18 | Blackvel | oh |
21:08.24 | Blackvel | could fix it |
21:08.40 | Blackvel | SAY NUMBER 12345 # |
21:08.43 | Blackvel | seems not working |
21:09.03 | Blackvel | what is in asterisk the default audio file? invalid? |
21:11.15 | *** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk) |
21:11.57 | zoa | there is no default audio file |
21:11.57 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
21:12.01 | *** join/#asterisk lesouvage (~lesouvage@cc341200-a.assen1.dr.home.nl) |
21:12.46 | Blackvel | any audio files which I can display to the caller? |
21:12.53 | Blackvel | like there is the "invalid" |
21:13.01 | zoa | lots of them |
21:13.08 | zoa | just look at the sounds directory |
21:13.15 | zoa | or the sounds.txt |
21:13.33 | Blackvel | I can believe it |
21:13.40 | Blackvel | my java application can control * :) |
21:13.44 | Blackvel | can't believe it |
21:15.13 | lesouvage | I have a x100p clown card. When I connect my phone and make a call after a while I hear "something terribly wrong, goodbuy". The output of the cli is Mar 26 16:08:26 NOTICE[1144]: app_dial.c:749 dial_exec: Unable to create channel of type 'Zap'. I disabled usb in the bios so there are no shared irg's. What should be my next step? |
21:15.49 | zoa | no support sorry |
21:15.59 | zoa | go buy one from digium and i will be happy to help you |
21:16.02 | zoa | and so will they |
21:19.09 | bjohnson | that's what you get for buying a clown card |
21:19.32 | bjohnson | did you get about 20 of them in a tiny little box? |
21:19.58 | lesouvage | zoa: Digium stops selling this card otherwise I would have bought one in there webshop. |
21:20.13 | zoa | they have tdm cards now |
21:20.28 | zoa | which are lot better |
21:20.34 | zoa | the x100ps suck |
21:20.56 | Blackvel | hm |
21:21.00 | Blackvel | what could be the reason for this: |
21:21.10 | Blackvel | I only receive one AGI message from * |
21:21.33 | Blackvel | but not all variables (multiple lines) in one read-call. I always have to read twice |
21:21.46 | Blackvel | agi_network: yes |
21:21.49 | Blackvel | that is the first |
21:22.45 | Blackvel | but the others I don't get in one read, e.g |
21:22.45 | Blackvel | agi://192.168.1.100 |
21:22.45 | Blackvel | agi_channel: SIP/100-2583 |
21:22.45 | Blackvel | agi_language: de |
21:23.05 | Blackvel | would be better to receive all in the first read |
21:26.35 | lesouvage | bjohnson: I just have one, just to test the concept of a mini asterisk box based on an epia m5000 motherboard. |
21:27.26 | *** join/#asterisk Juggie (agony@CPE00c049d9f271-CM014270110981.cpe.net.cable.rogers.com) |
21:27.45 | jakepdev | blackvel - read until the buffer is exhausted |
21:28.10 | jakepdev | or until a blank line might be safe |
21:28.15 | jakepdev | then post your commands |
21:29.21 | jakepdev | It is acceptable for TCP messages to be split among packets. You have to do your own parsing |
21:29.26 | KingMorse | trying again - anyone have any ideas why callerid is not working properly? I get some errors occasionally like those found here: http://www.mypastebin.com/?code=738791200 |
21:29.50 | KingMorse | CVS-HEAD, Digium TDM400 w/ 3 FXO modules |
21:30.24 | dant | KingMorse, where are you? |
21:30.34 | KingMorse | US |
21:30.53 | dant | Does it work on any of your lines? |
21:31.05 | KingMorse | none of them that I can tell |
21:31.25 | KingMorse | but callerid is active on the line, i checked with a regular phone |
21:31.29 | dant | Do you have callerID on the lines? |
21:31.30 | dant | ahh |
21:31.39 | dant | erm |
21:31.47 | KingMorse | i have usecallerid=yes, cidstart=ring, cidsignalling=bell |
21:32.02 | KingMorse | I've tried playing with txgain/rxgain, to no avail |
21:32.12 | KingMorse | I've tried turnning callprogress off |
21:32.26 | KingMorse | I've also tried with/without echo cancellation |
21:33.13 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
21:33.28 | dant | hmm |
21:34.00 | KingMorse | i've also tried immediate=no and a Wait(3) |
21:34.42 | KingMorse | the error messages are intermittent |
21:34.47 | KingMorse | they don't always show up, |
21:34.51 | dant | might help if someone else in the US who uses callerID on analogue lines could help |
21:34.52 | KingMorse | but CID never works regardless |
21:35.19 | dant | I'm in the UK, CID here is pretty different |
21:35.50 | KingMorse | ah. |
21:36.13 | dant | I'm not sure what to suggest next for your problem I'm afraid |
21:36.29 | KingMorse | that's quite alright, hopefully someone will see this and respond |
21:38.20 | *** join/#asterisk jhiver (~jhiver@AStDenis-103-2-3-232.w81-248.abo.wanadoo.fr) |
21:39.00 | jhiver | guys... can somebody send me a test email at jhiver@ykoz.net? I'm trying to figure out wether my mail server works... sorry 4 OT :: |
21:39.24 | *** join/#asterisk b0ef (~b0ef@062016141085.customer.alfanett.no) |
21:40.33 | b0ef | anyone tried using asterisk with jackplug?. I'm not getting it to select the jackplug device. I get snd_pcm_open failed: No such file or directory. I am able to use the jackplug device with other applications. |
21:41.15 | b0ef | jack is started as the same user as asterisk, btw |
21:41.47 | b0ef | I just don't get why it's not able to _see_ the jackplug device |
21:42.31 | Blissex | b0ef: if it works with an ALSA logical device name, it will work with any... |
21:43.36 | b0ef | Blissex: not sure what you mean; jackplug is a device defined in asound.conf |
21:44.45 | Blissex | b0ef: then if you can use 'jackplug' from any other ALSA app, it must work in Asterisk too. They all use ALSA lib, which interprets logical device names in exactly the same way for all apps. |
21:45.08 | Blissex | b0ef: note: try another ALSA app when logged in as the Asterisk login... |
21:45.27 | b0ef | Blissex: yes, I've done it and it works |
21:45.51 | b0ef | aplay -D jackplug foo.bar |
21:45.56 | Blissex | b0ef: logged in with the Asterisk login? |
21:45.56 | b0ef | as the asterisk user |
21:46.12 | b0ef | yes, I'm running it all under the asterisk user |
21:46.21 | Blissex | b0ef: then perhaps you have having hallucinations... :-) |
21:46.23 | *** join/#asterisk vlan (~iq@70-59-165-40.omah.qwest.net) |
21:46.40 | b0ef | asterisk is able to use the emu10k1 device specified in asound.conf aswell |
21:46.40 | cactus1 | is there anything i can do with asterisk @ home that would be neat before i get the card to connect to the pstn? |
21:46.45 | b0ef | but it won't use jackplug |
21:46.54 | *** part/#asterisk vlan (~iq@70-59-165-40.omah.qwest.net) |
21:49.19 | *** join/#asterisk vlan (~iq@70-59-165-40.omah.qwest.net) |
21:49.40 | cactus1 | b0ef is there anyway i can dial into my asterisk @ home box and call out to a FWD user or something to that effect? |
21:50.00 | b0ef | sure |
21:50.21 | cactus1 | would you recommend asterisk @ home? |
21:50.34 | b0ef | I don't even know what it is;) |
21:50.39 | *** part/#asterisk sezuan (sezuan@port-212-202-202-204.dynamic.qsc.de) |
21:50.42 | cactus1 | LOL |
21:50.46 | b0ef | , but asterisk can do that; it's a pbx |
21:51.18 | b0ef | asterisk @ home sounds like you got a asterisk box in your home;) |
21:51.26 | b0ef | s/a asterisk/an asterisk/ |
21:51.29 | cactus1 | heh |
21:51.30 | *** join/#asterisk goobster (goobster@c-67-168-105-166.client.comcast.net) |
21:51.31 | *** part/#asterisk goobster (goobster@c-67-168-105-166.client.comcast.net) |
21:51.34 | *** join/#asterisk goobster (goobster@c-67-168-105-166.client.comcast.net) |
21:51.46 | cactus1 | asterisk @ home is simplified and idiot proof compared to the full asterisk |
21:51.53 | bjohnson | asterisk@home is a CD iso that installs Centos and asterisk |
21:51.57 | b0ef | ahh |
21:52.02 | bjohnson | it is full asterisk |
21:52.21 | cactus1 | sorry bjohnson i worded it wrong |
21:52.22 | bjohnson | with the AMP gui .. which attracts idots .. but doesn't prevent them from screwing it up |
21:52.38 | b0ef | I'm no idiot so I don't need it;) |
21:52.56 | bjohnson | it's a quick way to get a asterisk box running |
21:53.12 | bjohnson | but unfortunately people try to use the AMP gui that comes with it |
21:53.13 | b0ef | yeah, but I got my own distro;), so I need to use that |
21:53.17 | vlan | Hi, how to play .gsm files ? |
21:53.26 | bjohnson | and then come here wanting us to solve their problems |
21:53.34 | b0ef | vlan: use sox |
21:53.35 | bjohnson | vlan: soxplay is one way |
21:53.41 | vlan | bjohnson: thanks |
21:54.00 | bjohnson | don't forget beef |
21:54.01 | bparker | anyone have any experience with Netlogic (VoIP Provider) |
21:54.04 | vidia | hi - where do change the email text that is sent along with voicemail messages? |
21:54.08 | bjohnson | err .. le boef |
21:54.12 | bjohnson | er b0ef |
21:54.17 | b0ef | ;) |
21:54.27 | b0ef | it's a recursive acronym |
21:54.39 | b0ef | b0ef 0wnz eats and fscks |
21:54.50 | b0ef | and it's sexadecimal;) |
21:58.15 | cactus1 | theres a bunch of people that are just dead weight sitting on this channel |
21:58.35 | b0ef | damn, it would be so nice to get asterisk running through jack |
21:58.46 | PatrickDK | dead weight? that's me |
21:58.47 | cactus1 | ? |
21:59.14 | b0ef | imagine the nice phonecalls;) |
21:59.19 | mgth | jack? |
21:59.22 | vidia | ? |
21:59.23 | file[laptop] | I never noticed but Voicepulse Connect upped their price on inbound DIDs... |
21:59.31 | mgth | to what? |
21:59.32 | Nugget | interesting. |
21:59.34 | file[laptop] | $11 |
21:59.37 | mgth | dang |
21:59.38 | cactus1 | http://cactus.dacnomm.com/calls/bh/product.wav <<caution may be innapropriate for some people |
21:59.44 | Nugget | they seem to have finally fixed the problems with area code 512 DIDs |
22:01.17 | vidia | anyone know where to change the email message that accompanys voicemail nessages?? |
22:01.18 | *** join/#asterisk booyeah23 (~afdas@cpe-24-175-29-253.houston.res.rr.com) |
22:01.33 | file[laptop] | vidia: voicemail.conf |
22:02.00 | booyeah23 | there are a lot of bugs dealing with transfers over bridged connections |
22:02.58 | *** join/#asterisk roamer323 (~sing@toronto-HSE-ppp4089458.sympatico.ca) |
22:03.48 | *** join/#asterisk phsdshft (~phsdshft@66.103.13.10) |
22:03.51 | roamer323 | ~trig |
22:05.24 | linagee | ~asterisk |
22:05.25 | jbot | well, asterisk is a PBX (Private Branch eXchange) and telephony toolkit. http://www.asterisk.org |
22:06.06 | Chuji | ~asstricks |
22:06.08 | jbot | i heard asstricks is #asstricks, the underground gay Asterisk channel. Be afraid, very afraid |
22:06.09 | linagee | lol |
22:06.16 | linagee | <PROTECTED> |
22:06.28 | IQ | loool |
22:06.43 | linagee | secret, underground, XOR-ed, channel. :) |
22:07.18 | b0ef | mgth: http://jackit.sf.net |
22:07.43 | booyeah23 | dtmf handling is also buggy over bridged connections |
22:09.26 | Essobi | chan_sip is coring on rtp.c in -head on me.. Anyone want to comment? |
22:09.59 | linagee | is there any way to have a grandstream make those annoying, "you left me off the phone too long and now i'm going to make annoying sounds" sound? |
22:11.06 | b0ef | I'm trying to get asterisk to work with jackplug. Both jackplug and emu10k1 are defined in /etc/asound.conf, but asterisk is only able to see and use the emu10k1 device. I'm able to aplay as the same user with jackplug. What could possibly be the problem here?. |
22:11.13 | booyeah23 | linagee: feedback |
22:11.29 | linagee | booyeah23: you mean there's actually a way? yuck. :) |
22:15.37 | booyeah23 | linagee: those phones are really cheap and crappy so its possible to get feedback from the two sides of the handset |
22:16.01 | linagee | booyeah23: grandstream is a cheap and crappy phone? :( |
22:16.07 | linagee | booyeah23: what would you suggest? |
22:16.37 | Moc | linagee, polycom, or it being told the new buisnes grandstream aint that bad, but nothing is confirm on that part |
22:16.37 | linagee | something that at least costs $300? |
22:17.05 | Moc | go with the polycom IP 600 then |
22:17.11 | *** join/#asterisk poli (~poli@200-168-30-125.dsl.telesp.net.br) |
22:17.34 | Shido6 | ? |
22:19.50 | Chuji | The new Grandstream I hear is OK |
22:20.20 | Chuji | Lots of people came back from VON talking about them |
22:20.39 | file[laptop] | I liked it. |
22:20.51 | *** join/#asterisk cbachman (~chatzilla@victory.ece.northwestern.edu) |
22:22.36 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
22:27.12 | tainted- | do variables span across contexts? |
22:27.13 | tainted- | or no |
22:27.31 | Essobi | umm |
22:27.37 | Essobi | depend on the variables.. |
22:28.11 | Essobi | There's global, and channel as far as I can remember |
22:28.27 | Essobi | So anyone else getting cores on -head? |
22:28.46 | *** join/#asterisk Secretive (~polarisx@c-67-161-5-149.client.comcast.net) |
22:28.53 | Secretive | Hey guys, anyone know what the hell this would mean: Mar 26 16:33:07 WARNING[2109]: chan_sip.c:739 retrans_pkt: Maximum retries exceeded on call 8332fb4f-61eedb89-2b337dd2@192.168.1.125 for seqno 106 (Non-critical Request) |
22:29.03 | file[laptop] | it couldn't send the packet |
22:29.22 | Secretive | How would you suggest I fix it? |
22:29.35 | file[laptop] | it's probably due to NAT |
22:29.46 | Secretive | Should I do some sort of Port Forwarding to my phone? |
22:30.00 | file[laptop] | do you have nat=yes in the the entry in sip.conf for the phone? |
22:30.14 | Secretive | Not sure it's in a database I believe. |
22:30.35 | file[laptop] | well if you don't have it set to yes, asterisk is going to send replies to the phone's internal address |
22:30.51 | file[laptop] | so I'd suggest turning it on... |
22:31.12 | file[laptop] | and depending on the NAT setup, you may need qualify turned on for it to receive calls |
22:31.18 | Secretive | Well -- Some guy setup this system and I can't get a hold of him. |
22:31.22 | file[laptop] | and yes, you may have to forward ports 5060 and 10000-20000 UDP |
22:31.59 | bkw_ | ok I gotta get this off my chest |
22:32.03 | bkw_ | EVERYONE CAN FUCK OFF... |
22:32.23 | bkw_ | people are such idiots at times |
22:32.31 | bkw_ | getting on to me for top posting.. OH FUCK THEM |
22:32.31 | Chuji | hmmm |
22:32.47 | file[laptop] | deep breaths bkw, deep breaths |
22:32.48 | Essobi | AND FUCK YOU, AND FUCK YOU, and you're cool, and IM OUT |
22:32.54 | bkw_ | haha |
22:32.56 | bkw_ | Essobi, ya that |
22:33.17 | file[laptop] | it's okay dear |
22:33.23 | bkw_ | next time i'll just let cdr_mysql be broken |
22:33.23 | Chuji | did you get critch'd ? |
22:33.26 | Essobi | D'oh.. Stick. |
22:33.51 | *** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released || http://www.theregister.co.uk/2005/03/16/asterisk_open_source_pbx/ || Quick everyone top post to the -dev and -biz list.. |
22:34.01 | Jer13261 | lol |
22:34.35 | bkw_ | thosee can do h323 |
22:34.36 | Chuji | amazing how telecom gear goes down in price |
22:34.44 | Essobi | :) |
22:34.50 | Chuji | that thing was 15,000 a few years ago |
22:34.59 | Jer13261 | $20??????????? |
22:35.17 | Essobi | You can find all kinds of stupid stuff on Ebay like that. |
22:35.31 | file[laptop] | they're cheap... |
22:36.07 | Chuji | 96-56k ports and 4T1 module |
22:36.15 | file[laptop] | bkw_: so, should I plan on attending cluecon? |
22:36.16 | bkw_ | they can do h323 and voice |
22:36.18 | bkw_ | NEXT!!! |
22:36.21 | bkw_ | file[laptop], yes |
22:36.29 | file[laptop] | but daddy do I have to |
22:36.37 | bkw_ | yes |
22:36.44 | file[laptop] | fine then |
22:36.51 | MikeJ[Jayden] | no, thats cheap: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=15635&item=4711941394&rd=1 |
22:37.04 | file[laptop] | I want my own little apx |
22:37.35 | bkw_ | why? |
22:37.46 | file[laptop] | bkw_: no reason, but it's nifty for smashing people over the head with |
22:37.47 | MikeJ[Jayden] | file[laptop], here you go http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=47064&item=7309411810&rd=1 :D |
22:38.09 | Chuji | bkw_ : Did you use max's when you worked at the ISP? |
22:38.48 | bkw_ | no |
22:38.51 | bkw_ | PM3's |
22:38.55 | bkw_ | I had friends that had a max |
22:39.09 | file[laptop] | bkw_: all your pbx are belong to me |
22:39.10 | *** join/#asterisk pimpsmart (~spam@cpe-24-175-29-253.houston.res.rr.com) |
22:39.22 | _Sam-- | i just threw away a bunch of pm3s |
22:39.47 | _Sam-- | bkw i didnt realize youre famous |
22:39.48 | _Sam-- | http://www.sineapps.com/news.php?rssid=378 |
22:39.58 | file[laptop] | ha |
22:40.03 | file[laptop] | _Sam--: I'm on there too ya know |
22:40.45 | pimpsmart | hey, anyone have trouble with dtmf when doing iax to iax? |
22:40.48 | file[laptop] | but enough about me, who are you and what do you want! how do you know my name... |
22:40.50 | pimpsmart | i'm using 1.0.6 |
22:41.01 | cjk | how cpu intensive is sip to iax and iax to sip translation |
22:41.22 | *** join/#asterisk shadebob (~shadebob@rnis-162-206-192-81.marocconnect.com) |
22:41.37 | file[laptop] | pimpsmart: do an iax2 debug and see if you're getting DTMF type packets... and if they're going back out |
22:42.28 | _Sam-- | bkw you're quoted "I would rather work on making the software more stable first. |
22:42.36 | _Sam-- | "....does that mean you feel the software isnt stable? |
22:42.45 | file[laptop] | okay let's not get into that |
22:42.47 | pimpsmart | file[laptop]: will do and report back |
22:43.29 | MikeJ[Jayden] | this is what I need: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=94879&item=5762790525&rd=1&ssPageName=WDVW |
22:44.02 | file[laptop] | MikeJ[Jayden]: crazy! |
22:44.52 | MikeJ[Jayden] | they are fun toys |
22:45.11 | pimpsmart | file[laptop]: no dtmf type packets |
22:45.27 | _Sam-- | what is the name of that software that is like ipswitchboard but done in flash? |
22:45.29 | file[laptop] | then it's not getting the DTMF from the other side... |
22:45.37 | file[laptop] | _Sam--: Flash Operator Panel |
22:45.41 | *** join/#asterisk crash3m (crash3m@crash3m.user) |
22:45.45 | _Sam-- | thankf forgot |
22:45.45 | file[laptop] | pimpsmart: SIP -> IAX -> whatever? |
22:45.52 | _Sam-- | f=s |
22:45.58 | Essobi | okay that's screwed up |
22:46.10 | pimpsmart | i am doing Cell phone->IAX->IAX->Cell |
22:46.11 | Essobi | I started * up and my 7960 starts ringing out of nowhere |
22:46.33 | file[laptop] | pimpsmart: well then, someone isn't sending you DTMF I'm afraid... |
22:46.35 | Chuji | My sipura does that |
22:46.44 | pimpsmart | iAX is nufone or voicepulse connect |
22:46.49 | pimpsmart | i tried both |
22:47.16 | file[laptop] | pimpsmart: there should be DTMF packets then... |
22:47.29 | file[laptop] | try from a different phone? |
22:47.30 | pimpsmart | yeah, that is what i thought |
22:47.49 | pimpsmart | tried sip->IAX->IAX->Cell |
22:48.07 | pimpsmart | but cell->IAX->SIP works fine |
22:48.15 | pimpsmart | could it be because of nat issues? |
22:48.33 | file[laptop] | I doubt it |
22:49.04 | file[laptop] | look on the first both where the call is coming for DTMF packets... |
22:49.08 | file[laptop] | er both = box |
22:50.21 | pimpsmart | for an iax2 debug, does it show other information other than the call setup? |
22:50.36 | file[laptop] | it'll show tons of info |
22:51.32 | file[laptop] | you should see something like this: (only two line paste) |
22:51.32 | file[laptop] | Tx-Frame Retry[000] -- OSeqno: 003 ISeqno: 004 Type: DTMF Subclass: 2 |
22:51.32 | file[laptop] | <PROTECTED> |
22:51.36 | file[laptop] | for transmitting DTMF |
22:51.42 | file[laptop] | and an Rx-Frame for receiving |
22:55.30 | pimpsmart | i still don't get that |
22:55.37 | pimpsmart | i get this though when i send dtmf when it works |
22:55.38 | pimpsmart | Mar 26 16:55:19 DEBUG[5434]: res_features.c:537 ast_bridge_call: Read from IAX2/NuFone@NuFone2/3 (1,49) |
22:56.14 | zoa | why am i not in sineapps ? :p |
22:56.31 | zoa | i also look gay on those last pictures |
22:56.44 | _Sam-- | thats ok zoa i still like ya :) |
22:56.47 | file[laptop] | zoa: you're sooooooo sexy |
22:56.52 | zoa | haha |
22:56.54 | pimpsmart | but when i did a transfer i got the dtmf from iax |
22:56.57 | pimpsmart | Rx-Frame Retry[ No] -- OSeqno: 056 ISeqno: 072 Type: DTMF Subclass: 0 |
22:56.57 | pimpsmart | <PROTECTED> |
22:57.32 | _Sam-- | big saturday night eh zoa? |
22:57.43 | zoa | nah just trying to stay awake |
22:57.48 | file[laptop] | pimpsmart: *shrug* |
22:57.55 | file[laptop] | hold me hold me HUG ME HUG ME |
22:57.58 | zoa | tsss |
22:58.02 | zoa | watch it sister |
22:58.11 | file[laptop] | just don't give me any germs |
22:58.18 | zoa | hehe lol |
22:58.30 | *** join/#asterisk r0d3nt|m (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
22:58.36 | file[laptop] | down below or up above :p |
22:58.41 | zoa | grr |
22:59.32 | zoa | could someone arrange me a picture of an iaxy wrapped in a condom for my next astricon speech about asterisk security ? :) |
23:00.09 | file[laptop] | what, can't do it yourself? |
23:00.24 | zoa | i dont have an iaxy |
23:00.35 | file[laptop] | Qwell: lol |
23:00.36 | zoa | that makes it kinda hard |
23:00.37 | Secretive | Anyway to have Playback or Background play a GSM file louder ? |
23:00.41 | file[laptop] | Qwell: I'm not even gonna ask why |
23:00.47 | Qwell | file[laptop]: good man |
23:01.37 | zoa | now get me that picture :p |
23:01.50 | Qwell | trust me, if I had a picture like that, you would've heard about it months ago :p |
23:03.13 | file[laptop] | alas, I have not an IAXy |
23:03.45 | Jovu | asterisk seems to shut itself down immediately, http://www.firenzee.com/full is the logfile but it doesnt seem to give any clues, i assume one of the sub threads is dying but i dont know how to debug that |
23:04.02 | Jovu | anyone have any ideas on this? i've been messing with this all day |
23:05.15 | *** join/#asterisk wildgoose (~edward@xunil.mail.wildgooses.com) |
23:05.33 | Shido6 | err |
23:05.35 | Shido6 | yeah |
23:05.40 | Shido6 | whats the log say? |
23:07.10 | *** part/#asterisk crash3m (crash3m@crash3m.user) |
23:08.39 | Jovu | the log is on http://www.firenzee.com/full |
23:08.45 | Jovu | doesnt seem to say anything usefull |
23:09.07 | Jovu | one minute its initializing, the next its shutting down |
23:12.43 | Essobi | Hmm.. something in -head rtp/chan_sip is out of wack. |
23:14.11 | poli | People, I need to find out the cadence parameters (prewink, preflash etc.) to get my X100P working with FXS signal from my PBX. Does anyone know some good source of information for that? thanks. |
23:14.48 | Shido6 | brb |
23:17.53 | Jovu | Essobi, i've tried with 1.0.6 and 1.0.7 stable versions too, and same problem.. i thought upgrading to -head might fix things |
23:19.17 | Essobi | that comment about -head wasn't meant for you |
23:19.49 | Essobi | there's something up with re-invite control in chan_sip -head |
23:20.52 | Essobi | Try uh... asterisk -cvvvvvggggg and you might see something useful |
23:23.27 | zoa | bkw, are you coming to astricon europe ? |
23:23.34 | zoa | file are you ? |
23:24.36 | file[laptop] | nope |
23:24.43 | zoa | damn |
23:25.18 | *** join/#asterisk Shido6 (~greg@d57-87-253.home.cgocable.net) |
23:25.59 | file[laptop] | Cluecon I can do |
23:26.16 | zoa | cluecon is so far away |
23:26.19 | file[laptop] | Astricon USA, I can do |
23:26.23 | file[laptop] | but Europe - HA |
23:26.50 | booyeah23 | Cluecon? |
23:27.10 | zoa | i would like to go to astricon eu, us, and von eu |
23:27.10 | NormAst | Can't do Europe. |
23:27.19 | zoa | and cluecon |
23:27.24 | zoa | but dont think i could do all of em |
23:27.38 | file[laptop] | if you don't show up to one I go to, I'll be sad :( |
23:27.39 | _Sam-- | where is astricon us? |
23:27.56 | zoa | atlanta again i think |
23:28.03 | _Sam-- | doesnt say on their page |
23:28.13 | *** join/#asterisk shidan (~shidan@CPE000e08eaf90e-CM014280007905.cpe.net.cable.rogers.com) |
23:28.14 | zoa | if you can go, go there, its really interesting |
23:28.15 | file[laptop] | that's because it is too far away |
23:28.24 | zoa | its les crap than the other things |
23:28.27 | zoa | its more hands on |
23:28.50 | zoa | most people speaking are not just talking about how great their company is |
23:28.55 | zoa | but talking about asterisk |
23:28.56 | SexyKen | I need to hire someone to write a GUI and setup an Asterisk server for me. |
23:29.13 | zoa | sexyken: find me in private :p |
23:30.02 | _Sam-- | what exactly is dCAP Certification |
23:30.07 | zoa | its a training |
23:30.15 | zoa | done by the people behind astricon |
23:30.20 | zoa | called olle and steve |
23:30.28 | zoa | they give a one week training on asterisk |
23:30.37 | _Sam-- | what does dpac stand for |
23:30.40 | zoa | and then they give you a certificate |
23:30.46 | zoa | good question :) |
23:30.51 | file[laptop] | dCAP, Digium Certified Asterisk Professional |
23:30.59 | zoa | ah voila |
23:31.13 | zoa | for the moment its just ipsando giving them |
23:31.48 | shidan | hey guys |
23:32.16 | zoa | hello there |
23:32.43 | file[laptop] | zoa zoa |
23:32.46 | zoa | yes yes ? |
23:32.53 | file[laptop] | zoa zoa zoa |
23:33.02 | file[laptop] | or rather, Joachim! |
23:33.07 | poli | In zapata.conf faxdetect=incoming means Asterisk will detect fax signal coming from an incoming connection or the "incoming" is the context? |
23:33.45 | zoa | its on incoming |
23:33.52 | zoa | not the context |
23:33.56 | file[laptop] | zoa: you are very silly |
23:33.57 | *** join/#asterisk Darwin35 (~Darin@c-24-3-226-147.client.comcast.net) |
23:33.59 | zoa | context would be the defaultcontext stuff |
23:34.03 | zoa | thank you file :) |
23:34.08 | shidan | does anyone know of a ser live cd thats free by any chance? |
23:34.11 | file[laptop] | but a great guy regardless |
23:34.26 | _Sam-- | damn zoa, is file your biggest fan, or you have others? |
23:34.45 | file[laptop] | be careful, zoa can tackle you and win |
23:34.48 | elriah | Anyone running debian with the 1.0.5 asterisk package? |
23:34.48 | poli | zoa, If I can't select the context a fax call is going to, what is faxdetection good for? |
23:34.51 | file[laptop] | he can flip you! |
23:35.03 | jeffik | Shidan: Chk you pvt |
23:35.11 | _Sam-- | i will not be in any positions where zoa will be flipping and/or tackling me. |
23:35.11 | shidan | oh ok ;) |
23:35.17 | _Sam-- | but thanks for the heads up. |
23:35.42 | zoa | neither will file, but he keeps dreaming :p |
23:35.49 | file[laptop] | ha |
23:42.13 | IQ | Hi, whats the command syntax to play .gsm using sox? |
23:44.00 | Essobi | IQ: sox is a player too? I thought it was just amixer. ;) |
23:44.32 | IQ | Essobi: so thats why its not playing :P ... |
23:44.44 | IQ | Essobi: what do u use to play - or u convert it into .wav first ? |
23:44.49 | Essobi | <PROTECTED> |
23:45.02 | Essobi | quicktime will play GSM. |
23:45.26 | Essobi | but that's on windows.. no idea for anything else.. |
23:46.06 | IQ | I'm using Fedora |
23:46.16 | Essobi | So anyone else having trouble with -head SIP, reinvite/monitor/ast_channel_bridge? |
23:46.24 | Essobi | IQ: No idea. |
23:47.02 | *** join/#asterisk JerJer[mobile] (proxyuser@adsl-66-143-63-132.dsl.ksc2mo.swbell.net) |
23:47.37 | Essobi | I got a repeatable core in -head where ast_rtcp_read is reading a null pointer and coring. |
23:48.09 | shidan | just out of curiosity has anyone here tried compiling ser on windows using cygwin |
23:56.03 | JerJer[mobile] | shidan: why in hell would anyone want to do a thing like that? |
23:56.17 | file[laptop] | JerJer[mobile]: because Windows is the best server OS!!! |
23:56.22 | file[laptop] | the ads told me so so it must be true!!! |
23:56.28 | shidan | haha |
23:56.37 | shidan | because when im at work i have to run windows |
23:56.44 | shidan | the corporate image |
23:56.49 | shidan | workstation image |
23:56.56 | shidan | and i want to work on my ser scripts |
23:57.11 | FuriousGeorge | i think im beginning to get this: true or false: at the very least i will need a context for incomming and outgoing calls |
23:58.23 | shidan | im running the cygwin port for asterisk and it works fine for dev purposes |
23:58.37 | JerJer[mobile] | Essobi: http://bugs.digium.com |