| 00:00.22 | L|NUX | i am trying to call from one * to anoter * but its saying the personal you are calling is not available please try again |
| 00:00.35 | L|NUX | getting this one console |
| 00:00.36 | L|NUX | Apr 20 18:56:04 WARNING[9426]: chan_sip.c:862 retrans_pkt: Maximum retries exceeded on call 9c1b9b2282588f71 for seqno 1 (Critical Response) |
| 00:00.36 | L|NUX | Apr 20 18:56:12 WARNING[9426]: chan_sip.c:862 retrans_pkt: Maximum retries exceeded on call 9c1b9b2282588f71 for seqno 1 (Critical Response) |
| 00:00.36 | L|NUX | Apr 20 18:56:28 WARNING[9426]: chan_sip.c:862 retrans_pkt: Maximum retries exceeded on call 9c1b9b2282588f71 for seqno 1 (Critical Response) |
| 00:00.54 | Weezey | Linux; IAX2 or SIP? |
| 00:00.57 | L|NUX | SIP |
| 00:00.59 | Storhost | what's the best/easiest gui interface for asterisk ? |
| 00:01.04 | L|NUX | AMP |
| 00:01.08 | Storhost | thank for the MeetMe link... |
| 00:01.14 | L|NUX | Storhost : AMP |
| 00:01.36 | Weezey | Storhost; no problem, there's Conference() too. |
| 00:01.47 | L|NUX | Weezey : any idea |
| 00:01.48 | L|NUX | ? |
| 00:01.49 | pgpkeys | easiest maybe, best, no. best is for you to learn the conf files |
| 00:01.52 | Dishwasha | So is there no way in Asterisk to script a SIP header rewrite? |
| 00:01.54 | TechDawg | I finally have the two boxes running Asterisk/Zaptel 1.0.7 and the hardware being recognized. Now I have no clue where to go from this point. What I'm wanting to do is have an incoming PSTN call routed to the other Asterisk box and out the FXS port. Any help would be appreciated. |
| 00:02.12 | Weezey | Linux: you have two contexts? one for in and one for out? |
| 00:02.19 | L|NUX | yeah |
| 00:02.30 | Weezey | are you registering one to the other? |
| 00:02.36 | L|NUX | well |
| 00:02.45 | L|NUX | weezey : i have one context = sip |
| 00:02.52 | L|NUX | which have all incoming sip to sip |
| 00:03.02 | L|NUX | and when i try to call from another sip service to my server |
| 00:03.18 | L|NUX | i.e 1002@e-maili.com it say not availavble |
| 00:03.22 | L|NUX | available |
| 00:03.55 | Weezey | in sip.conf what's the context for your outgoing to the other box? |
| 00:04.15 | Weezey | I would do Dial(SIP/context/${EXTEN}) |
| 00:04.32 | SimonR | has anyone use ganglia instead of nagios? |
| 00:04.34 | L|NUX | well its in my extenstion.com |
| 00:05.05 | Weezey | right, but you said in sip.conf you have two contexts, one for in and one for out. |
| 00:05.07 | *** join/#asterisk ManxPower (~eric@dsl-209-205-172-111.i-55.com) |
| 00:05.10 | *** join/#asterisk JonM (~JonM@pc-68-118-196-199.will.ct.charter.com) |
| 00:05.17 | Weezey | Manx. |
| 00:05.25 | L|NUX | Weezey : right now i have only one |
| 00:05.29 | L|NUX | don't have two context |
| 00:05.31 | JonM | hey guys |
| 00:05.32 | dmccollum | Does anyone know if AMP works with the Asterisk Realtime in HEAD? |
| 00:05.32 | Weezey | have two |
| 00:05.58 | L|NUX | can you try to dial at 1002@e-maili.com |
| 00:06.10 | Weezey | sec |
| 00:06.13 | L|NUX | k |
| 00:06.38 | Sedorox | how would you dial that from a hard phone? or would you have to put it in your extentions? |
| 00:06.52 | JonM | looking to find a list of all the sound files that come with asterisk. Anyone know of any? And also what is a good resorce for dial plans |
| 00:06.52 | L|NUX | well i am dialing it from softphone atm |
| 00:06.53 | L|NUX | :) |
| 00:06.59 | ManxPower | Ugh. I hate people. |
| 00:07.09 | Nugget | Me too. |
| 00:07.11 | Sedorox | ah |
| 00:07.14 | JonM | third that |
| 00:07.15 | L|NUX | ~google asterisk sound |
| 00:07.26 | L|NUX | http://www.voip-info.org/wiki-Asterisk+sound+files |
| 00:07.29 | L|NUX | this one |
| 00:07.32 | dmccollum | I'm an Animal, or so a few women have told me. :P |
| 00:07.34 | Weezey | Linux: Got SIP response 404 "Not Found" back from 72.20.10.51 |
| 00:07.40 | L|NUX | hmm |
| 00:07.41 | L|NUX | what |
| 00:07.49 | Weezey | sounds like that box doesn't know where 1001 is. |
| 00:07.55 | JonM | thank you |
| 00:07.55 | L|NUX | oh |
| 00:07.56 | L|NUX | shit |
| 00:08.01 | L|NUX | its 1002@e-maili.com |
| 00:08.09 | *** join/#asterisk dizzydiffi (dizzydiffi@adsl-70-240-211-145.dsl.hstntx.swbell.net) |
| 00:08.23 | dizzydiffi | hello |
| 00:08.27 | JonM | Weezy i had that same error last night |
| 00:08.40 | Weezey | brb dinner. |
| 00:08.48 | Sedorox | L|NUX: you did say 1002 before :pp; |
| 00:08.55 | L|NUX | yupz |
| 00:08.55 | dizzydiffi | hello |
| 00:09.01 | L|NUX | <L|NUX> can you try to dial at 1002@e-maili.com |
| 00:09.01 | L|NUX | <Weezey> sec |
| 00:09.06 | dizzydiffi | anyone got oh323 to work with sip |
| 00:09.07 | Sedorox | hehe |
| 00:09.22 | L|NUX | dizzydiffi : yupz i did |
| 00:09.58 | dizzydiffi | great |
| 00:09.58 | dizzydiffi | i have a question |
| 00:09.58 | L|NUX | yes |
| 00:09.58 | dizzydiffi | i got sip to oh323 to work through the gnugk |
| 00:09.58 | L|NUX | but donno how to use it after compiling :) |
| 00:09.59 | L|NUX | hehe |
| 00:10.12 | dizzydiffi | but i cant get h323 to sip |
| 00:10.13 | L|NUX | dizzydiffi : i am n00b just installed it :) |
| 00:10.25 | L|NUX | ask some one else may be help you |
| 00:10.29 | dizzydiffi | okay |
| 00:10.31 | dizzydiffi | thanks |
| 00:10.37 | L|NUX | if you need help in compiling then i would help full :) |
| 00:10.50 | dizzydiffi | i got it working |
| 00:11.01 | dizzydiffi | i can make sip to h323 calls |
| 00:11.16 | L|NUX | hmm |
| 00:11.25 | dizzydiffi | ya with gnugk |
| 00:11.30 | L|NUX | can you call me on my sip for testing 1002@e-maili.com |
| 00:11.35 | Storhost | <dizzydiffi> i can make sip to h323 calls <- define that please |
| 00:11.43 | Storhost | does that mean pc to telephone? |
| 00:11.48 | dizzydiffi | okay with my sip phone |
| 00:11.52 | L|NUX | yeah |
| 00:11.54 | Storhost | hmm. |
| 00:11.57 | dizzydiffi | i call the h323 phone |
| 00:12.03 | dizzydiffi | through the gnugk |
| 00:12.06 | Storhost | h323 being a "land line" phone, right? |
| 00:12.16 | Storhost | or is h323 the software phone |
| 00:12.18 | dizzydiffi | no voip protocol |
| 00:12.21 | Storhost | ok |
| 00:12.30 | Storhost | yeah, duh.. h323... netmeeting uses it |
| 00:12.31 | Storhost | sorry. |
| 00:12.37 | dizzydiffi | ya ya |
| 00:12.40 | dizzydiffi | thats alright |
| 00:12.56 | dizzydiffi | so no one in here got it work as far as i have |
| 00:12.58 | Storhost | can Asterisk do software phone to landline? |
| 00:13.10 | L|NUX | when i dialing from another service got this on CLI |
| 00:13.11 | L|NUX | Apr 20 19:07:39 WARNING[9426]: chan_sip.c:862 retrans_pkt: Maximum retries exceeded on call c109e743a727c70f for seqno 1 (Critical Response) |
| 00:13.55 | dizzydiffi | ya it can |
| 00:14.03 | Storhost | nice |
| 00:14.18 | Storhost | what about the other way around? Land line to software |
| 00:14.35 | L|NUX | well |
| 00:14.37 | L|NUX | configure DID |
| 00:14.39 | dizzydiffi | it work both ways pstn network |
| 00:14.44 | L|NUX | Direct Inward Dialing |
| 00:15.10 | *** join/#asterisk TheEmperor (user@218.111.49.253) |
| 00:15.36 | dizzydiffi | how come no one has h323 working |
| 00:16.02 | dizzydiffi | with asterisk |
| 00:16.21 | Sedorox | its hard... lol.. at least from what I've seen |
| 00:16.27 | TheEmperor | dizzydiffi: i do |
| 00:16.31 | dizzydiffi | oh yea |
| 00:16.38 | dizzydiffi | what have you got so far the Emperor |
| 00:16.40 | Weezey | back. |
| 00:16.41 | TheEmperor | dizzydiffi: but i am having trouble with the quality.. |
| 00:16.41 | dizzydiffi | i got it working |
| 00:16.46 | dizzydiffi | oh yea |
| 00:16.51 | dizzydiffi | quality is fine for me |
| 00:16.51 | TheEmperor | sound quality is not so good |
| 00:16.59 | TheEmperor | what codec have you been using? |
| 00:17.11 | Weezey | diz: I have h323 working just fine |
| 00:17.17 | TheEmperor | my gatekeeper only supports g729 |
| 00:17.28 | TheEmperor | what codecs you guys been using? |
| 00:17.32 | L|NUX | :> |
| 00:17.36 | L|NUX | i am using g729 |
| 00:17.37 | Weezey | ulaw and 726 |
| 00:17.51 | *** part/#asterisk SimonR (~SimonR@Toronto-HSE-ppp3736980.sympatico.ca) |
| 00:17.52 | dizzydiffi | the only problem i have cant call sip phone |
| 00:17.55 | TheEmperor | yeah, even with 726 i am having problems... |
| 00:18.16 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 00:18.29 | Weezey | Linux; how did you get 729? You buy a license? |
| 00:18.31 | TheEmperor | i always get this error message : |
| 00:18.35 | TheEmperor | Apr 21 08:15:30 WARNING[1247802432]: samples/codec_g729.c:217 g729tolin_framein: Received a G.729 frame that was 4 bytes from RTP |
| 00:18.46 | TheEmperor | do you guys know what the problem is? |
| 00:19.05 | dizzydiffi | did you get the calls from h323 to sip |
| 00:19.06 | *** join/#asterisk mindamp (~mindamp@AC8ACF39.ipt.aol.com) |
| 00:19.24 | Weezey | diz: yeah |
| 00:19.35 | dizzydiffi | how did you do that |
| 00:19.41 | TheEmperor | any idea guys? |
| 00:20.00 | dizzydiffi | i can only call sip to h323 |
| 00:20.20 | TheEmperor | I am using IAX2 to call to h323 |
| 00:20.29 | Weezey | diz: what do you see on the console when you go h323 to sip? |
| 00:20.59 | timecop | h323 fucking fails it |
| 00:21.07 | dizzydiffi | well using gnguk i get a busy tone |
| 00:21.21 | TheEmperor | any help? :( |
| 00:21.52 | dizzydiffi | what the you do Weezey |
| 00:21.56 | Weezey | timecop: that's what your console says? You must have the vulgar version. |
| 00:22.26 | Weezey | diz: from the 3com nbx, I just dial the IP and the extension of my * box and it works. |
| 00:22.29 | *** join/#asterisk ropeguru (~ropeguru@fw.ropeguru.com) |
| 00:22.39 | dizzydiffi | do you use the gnugk |
| 00:23.23 | timecop | i got some shitty chinese h323 phone and h323 provider |
| 00:23.27 | timecop | and asterisk fails to use htem |
| 00:23.30 | timecop | cant find compatible codecs |
| 00:23.37 | timecop | even though I told the chinese to make it 711u only |
| 00:23.41 | Weezey | you've tried ulaw? |
| 00:23.44 | TheEmperor | timecop: i think that might be my problem.. |
| 00:23.48 | Weezey | hmm. |
| 00:23.50 | timecop | TheEmperor: h.323 debug |
| 00:23.58 | timecop | and when you place a call, notice if it says -- no compatible codecs for <foo> |
| 00:24.12 | dizzydiffi | hey Weezey what GK do you use Gnugk |
| 00:24.25 | Weezey | none |
| 00:24.32 | TheEmperor | timecop: i get this message |
| 00:24.35 | TheEmperor | WARNING[1264579392]: samples/codec_g729.c:217 g729tolin_framein: Received a G.729 frame that was 4 bytes from RTP |
| 00:24.43 | TheEmperor | timecop: any idea? |
| 00:24.58 | timecop | oh |
| 00:24.59 | timecop | thats normal |
| 00:25.15 | timecop | as in i dont think it should affect anything. |
| 00:25.20 | dizzydiffi | do you use Oh323 |
| 00:25.48 | TheEmperor | really? |
| 00:26.31 | Weezey | sounds like you don't have a 729 license or something. |
| 00:26.43 | dizzydiffi | anyone used Openh323 + gnugk with H323 + SIP phones |
| 00:26.51 | timecop | Weezey: if he didnt, it wouldnt work at all. |
| 00:26.54 | TheEmperor | Weezey: yeah...i should get one |
| 00:26.55 | Weezey | oh |
| 00:26.55 | dizzydiffi | i know someon has done this |
| 00:26.58 | timecop | uh? |
| 00:27.08 | timecop | TheEmperor: you dont have a 729 license? |
| 00:27.22 | timecop | then why did you enable g729? |
| 00:28.07 | Weezey | timecop: it's cool. |
| 00:28.24 | Weezey | I gotta get me a license or two. |
| 00:28.36 | dizzydiffi | all i wanna kknow is how to make a call with a h323 phone |
| 00:28.49 | *** part/#asterisk Storhost (~rewt@adsl-210-7-169.mco.bellsouth.net) |
| 00:28.51 | timecop | pick it up? |
| 00:28.52 | dizzydiffi | the gk doesnt want to route the call to the asterisk gateway |
| 00:28.56 | dizzydiffi | i need help |
| 00:28.59 | *** join/#asterisk chris_d (~chris@66.88.142.66.ptr.us.xo.net) |
| 00:29.07 | Weezey | dizzy: is it firewalled? |
| 00:29.11 | dizzydiffi | no |
| 00:29.13 | timecop | dial(h323/lol@ip) or something. |
| 00:29.32 | dizzydiffi | but how do you set up the gnugk |
| 00:29.50 | dizzydiffi | i dont understand the alisa thing |
| 00:29.58 | timecop | why are you using a gatekeeper thingy anyway. |
| 00:30.07 | Weezey | dizzy: mine works just fine without a gatekeeper |
| 00:30.20 | Weezey | </quit> |
| 00:31.01 | dizzydiffi | well to use openh323 |
| 00:31.12 | dizzydiffi | how did you do that weezey |
| 00:31.35 | Weezey | Uh, I just installed oh323, then I dial the call in or out. |
| 00:31.38 | Weezey | it just works. |
| 00:32.10 | timecop | how does one go about obtaining G723? |
| 00:32.16 | dizzydiffi | oh |
| 00:32.27 | dizzydiffi | <PROTECTED> |
| 00:32.36 | Weezey | nope |
| 00:32.42 | dizzydiffi | huh |
| 00:33.14 | ManxPower | ARGH! The city I'll be in tomorrow doesn't have any cab companies. It's plenty big enough to have a cab company, but they just don't want people without a car to be there. They don't have any sidewalks either. |
| 00:33.39 | timecop | where the hell can I get g723 |
| 00:33.40 | timecop | ? |
| 00:33.43 | timecop | or pay for it even |
| 00:33.45 | Weezey | where you goin'? |
| 00:33.57 | ManxPower | Weezey, suburb of New Orleans |
| 00:34.02 | *** join/#asterisk DaLion (~DaLion@HSE-QuebecCity-ppp3496739.sympatico.ca) |
| 00:34.11 | dizzydiffi | i guess in your oh323 conf you didnt set the gatekeeper |
| 00:34.26 | Weezey | nope |
| 00:34.33 | ManxPower | Of course, this is the same city that has a cop that sits on the side of one of the main roads into town to stop the non-whites. |
| 00:34.33 | dizzydiffi | agh! |
| 00:34.36 | dizzydiffi | what did you do |
| 00:35.01 | ManxPower | Sometimes The South really annoys me. |
| 00:35.13 | timecop | does anyone know? |
| 00:35.14 | timecop | wtf.. |
| 00:35.30 | ManxPower | timecop, You can't. You can only get illegal G723.1 |
| 00:35.34 | Weezey | timecop; I dunno man, can't find it. |
| 00:35.43 | dmccollum | That's also the state that considers a piece of tape over the straw hole to be a closed container. |
| 00:36.08 | timecop | ManxPower: well, I dont care at this point I want to see if thi sshit will work wiht h323. i see codec_g723_1.c in the source code which includes 723b/ dir or something |
| 00:36.11 | ManxPower | dmccollum, well there are a FEW cool things about this area. |
| 00:36.13 | timecop | where hte hell do I get this? |
| 00:36.33 | ManxPower | ~google site:lists.digium.com intel g723.1 |
| 00:36.39 | Weezey | timecop: http://lists.digium.com/pipermail/asterisk-users/2004-April/044445.html |
| 00:37.25 | *** join/#asterisk riksta (~rick@81-178-209-106.dsl.pipex.com) |
| 00:37.40 | MarkS__ | does anyone have experience installling festival and can help a n00b? pm me if possible |
| 00:38.14 | dmccollum | Does anyone know if AMP works with the Asterisk Realtime? |
| 00:38.25 | ManxPower | dmccollum, I doubt it |
| 00:38.27 | MarkS__ | shido6 - so you all arent taking any new customers? |
| 00:38.39 | timecop | well microsoft must have really paid a lot for 723 |
| 00:38.42 | timecop | to include it in netmeeting |
| 00:38.53 | Weezey | Mark: not until their new site is done. |
| 00:39.08 | *** join/#asterisk zione (~zione@62-101-126-208.fastres.net) |
| 00:39.10 | ManxPower | MarkS__, The only way to install Festival is to read the docs VERY CAREFULLY |
| 00:39.26 | dmccollum | AMP uses a database table to store the configs before it writes it out to the text files doesn't it? If so I bet I could setup realtime to use those same tables. |
| 00:39.32 | timecop | ManxPower: so this? http://www.readytechnology.co.uk/open/g723.1/ |
| 00:39.33 | MarkS__ | ahhh |
| 00:39.55 | MarkS__ | no easy way thru emerge/portage? |
| 00:40.38 | timecop | wtf |
| 00:40.43 | timecop | i dont understand waht hte hell that page is talking about. |
| 00:40.48 | timecop | they have a diff for what? |
| 00:40.54 | timecop | its too small to be the codec... |
| 00:41.15 | PBXtech | its just a diff :) |
| 00:41.40 | timecop | i guess I hsould read the install doc. |
| 00:42.01 | DaLion | u guys remmeber seing a script that dumps realtime to .conf ? i cant seem to find it back.. but saw something about it somewhere |
| 00:43.03 | ManxPower | timecop, Yes. That scum of the earth. |
| 00:43.04 | Weezey | timecop: it's a bitch, I started the one for 729, I think I needed to be running it on a P4 for it to work. |
| 00:43.37 | timecop | jesus christ what the ipp shit is like 140megs |
| 00:43.39 | timecop | i hope this is worth it |
| 00:43.58 | PBXtech | you sure do complain alot about pirated software |
| 00:44.37 | Weezey | PBXtech: it's educational. |
| 00:44.38 | timecop | what hte hell is IPP anyway |
| 00:44.40 | timecop | intel what what? |
| 00:44.52 | Weezey | Intel Performed Practically all the work. |
| 00:44.52 | Nugget | get down with IPP (yeah you know me) |
| 00:44.55 | Sedorox | internet prinint protocol |
| 00:44.56 | Sedorox | ? |
| 00:45.21 | timecop | if this shit requires a p4, i'm out i'm running on a p3 |
| 00:45.32 | Weezey | now I'm naughty by nature, not cuz I hate ya |
| 00:46.03 | Weezey | timecop: it wouldn't work for me and we narrowed it down to that same problem, I was going to try P4 next. |
| 00:47.24 | ManxPower | Weezey, The ONLY entity that can grant you an educational license for the PATENTED tech, is the patent holder. Hell, even Intel's own readme says you need to obtain a license for the codecs. |
| 00:47.35 | timecop | so im wondering whats the 723 thats included in asterisk source |
| 00:47.42 | timecop | or rather not included, but the dir |
| 00:48.00 | Weezey | ManxPower: really? huh, so it didn't work because of that, that's good. |
| 00:48.21 | ManxPower | timecop, I suspect it was a placeholder until Digium could license G723.1, but of course the G723.1 patent holders want $30,000 just to talk to you. |
| 00:48.22 | Weezey | I just wanted to try a g729 call or two to make sure it was worth spending money on. |
| 00:48.30 | timecop | it is. |
| 00:48.36 | ManxPower | Weezey, no, it should work, it's just not legal 8-) |
| 00:48.37 | timecop | so just buy it. |
| 00:49.04 | Weezey | timecop: the only reason I'm skeptical is that 726 is compressed and it sounds like my balls. |
| 00:49.10 | ScythelX | anyone using odbc on freebsd for res_odbc |
| 00:49.17 | Weezey | 8k sounds too good to be true. |
| 00:49.57 | timecop | well i have 2 channels of it |
| 00:49.58 | timecop | it sounds good. |
| 00:50.20 | MarkS__ | in voicemail.conf for the field mailcmd can i put a remote mail server? ||| the current is ;mailcmd=/usr/sbin/sendmail -t but i dont have or want a mail server on the server which asterisk is currently hosted on!! |
| 00:50.26 | Weezey | k, well, I guess I should just buy two and see for myself. |
| 00:51.52 | timecop | hm |
| 00:51.56 | ManxPower | MarkS__, No. Asterisk does not have a buit in SMTP client. It just pipes the message to a local executable and lets the executable deal with it |
| 00:51.58 | timecop | looks like the have p3 build flagsin the makefile |
| 00:52.01 | timecop | so i guess itlll work |
| 00:52.13 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
| 00:52.23 | ManxPower | Weezey, 8k is just for audio, you still have lots of UDP overhead. |
| 00:52.26 | MarkS__ | fuck |
| 00:52.30 | ManxPower | 24k of overhead, I think. |
| 00:52.32 | MarkS__ | well.. any alternative ways to do it? |
| 00:52.39 | timecop | MarkS__: so, write a perl script as /usr/sbin/sendmail |
| 00:52.47 | MarkS__ | hell i dont know perl |
| 00:52.48 | ManxPower | MarkS__, install a probgram that IS an SMTO client |
| 00:53.00 | MarkS__ | but any script would do, what would it be called, so i can look one up on google |
| 00:53.01 | timecop | install something like exim |
| 00:53.03 | timecop | and make it only local |
| 00:53.30 | ManxPower | I install a local only postfix server on my Asterisk servers |
| 00:53.36 | timecop | or that. |
| 00:53.41 | timecop | exim > postfix though |
| 00:53.53 | ManxPower | Mandrake defaults to local only for it's default postfix install I think |
| 00:54.30 | MarkS__ | huh |
| 00:54.31 | *** join/#asterisk MrEntropy (~entropy@170.003.dsl.sa.iprimus.net.au) |
| 00:54.32 | MrEntropy | yo |
| 00:54.53 | MarkS__ | so there isnt just a perl script i can find on google that will do the job? |
| 00:55.09 | Weezey | timecop: I used the PIII and it didn't work, PII also. |
| 00:55.27 | MrEntropy | i just booted windows messenger with the hope of enablind the dial pad. Upon launch it kept bothering me to update it, did the update remove the dialpad, because even after the regkey edit i can't get it? |
| 00:55.27 | timecop | he |
| 00:55.28 | timecop | h |
| 00:58.39 | *** join/#asterisk tzanger (~tzanger@38.116.194.42) |
| 00:58.47 | Weezey | tzang! |
| 00:58.53 | timecop | what hte fuck.. |
| 00:59.04 | timecop | it untars into a huge .exe which finally installs into a RPM |
| 00:59.10 | timecop | fucking intel |
| 00:59.18 | timecop | now what the hell am I going to do wiht a RPM |
| 00:59.31 | Weezey | follow the instructions? |
| 01:02.04 | MrEntropy | timecop: are you installing the g729 edu libs? |
| 01:02.13 | ManxPower | ARGH! My fave type of hotel room isn't available! |
| 01:02.21 | timecop | nah |
| 01:02.23 | timecop | I paid for 729 |
| 01:02.29 | timecop | i need to see if 723 is hte problem h323 isnt wokrign with this chink shit |
| 01:03.00 | Weezey | ManxPower: you're having a great day. |
| 01:03.26 | ScythelX | whats the best linux distro to run asterisk |
| 01:03.26 | MrEntropy | but you're talking about the intel libs l_ipp_ia32? |
| 01:03.32 | ScythelX | i hate redhat, but im a freebsd user |
| 01:03.51 | ManxPower | Weezey, It's the start of Jazz Fest this weekend. |
| 01:03.59 | Weezey | ScythelX, why don't you run FreeBSD? |
| 01:04.02 | ManxPower | I always try to get a room with the Spa Tub. |
| 01:04.03 | timecop | MrEntropy:yes |
| 01:04.06 | timecop | fucing around with that now. |
| 01:04.15 | *** join/#asterisk hypa7ia (~leigh@toronto-HSE-ppp4062725.sympatico.ca) |
| 01:04.25 | ScythelX | Weezey: having problems getting res_odbc to compile and the asterisk-addons |
| 01:04.43 | timecop | still wondering what version of 723.1 is referenced by the asterisk installer |
| 01:04.51 | Weezey | ScythexlX: so you just give up |
| 01:04.54 | MrEntropy | timecop: i used rpm2cpio to extract that rpm |
| 01:05.49 | ScythelX | no, but i just dont think asterisk is going to work well with freebsd |
| 01:06.05 | *** join/#asterisk tengulre (~tengulre@61.185.238.166) |
| 01:06.22 | Weezey | works fine |
| 01:06.42 | ScythelX | and your using res_odbc |
| 01:06.47 | ManxPower | Spa Tub == Bliss |
| 01:07.00 | Weezey | ScythelX: yeah |
| 01:07.22 | Weezey | like I said, I just googled the error it gave me and there was the fix, then it all worked. |
| 01:07.23 | Sedorox | ScythelX: what problems you having? all three of my * boxes are fbsd... |
| 01:07.40 | Weezey | Sedorox: that's what I like to hear. |
| 01:08.06 | Sedorox | the only problem I had is a error in Zaptel that would lock the box solid.. but thats been fixed in 0.9 |
| 01:09.06 | ScythelX | well asterisk detects the unixodbc |
| 01:09.13 | ScythelX | but i get an error on compile |
| 01:09.19 | ScythelX | its looking for lodbc |
| 01:09.29 | ScythelX | and i even installed the C++ libs |
| 01:09.39 | ScythelX | gcc -shared -Xlinker -x -o res_odbc.so res_odbc.o -lodbc |
| 01:09.39 | ScythelX | /usr/bin/ld: cannot find -lodbc |
| 01:09.53 | Sedorox | hmmm |
| 01:10.06 | Sedorox | haven't seen that |
| 01:10.12 | *** join/#asterisk newmember (~newmember@S010600a0c93dce87.cg.shawcable.net) |
| 01:10.43 | Weezey | Sedorox: he just doesn't know how to use google... |
| 01:10.49 | Weezey | ScythelX: http://lists.digium.com/pipermail/asterisk-bsd/2005-January/000526.html |
| 01:11.22 | Weezey | when googling, for - something you must put it in quotes like so: /usr/bin/ld: cannot find "-lodbc" asterisk |
| 01:11.39 | Weezey | otherwise it finds everything but lodbc |
| 01:11.45 | ManxPower | screw it. It's time for intoxicants |
| 01:11.52 | ManxPower | it IS 4/20 after all. |
| 01:11.59 | Weezey | ideed |
| 01:13.34 | Sedorox | lol |
| 01:13.55 | Sedorox | and here I thought 4/20 was only a HS thing... oi |
| 01:14.15 | ScythelX | ok gonna try it |
| 01:14.16 | ScythelX | thanks |
| 01:14.24 | *** join/#asterisk lashkar (~nobody@cookeville-68-112-64-5.midtn.chartertn.net) |
| 01:16.06 | Micc | man, I can't wait to get my nufone.net numbers. |
| 01:16.24 | Micc | the ping time to their server is twice as fast as broadvoice. |
| 01:16.45 | timecop | hm |
| 01:16.46 | timecop | it built. |
| 01:17.21 | tengulre | Hi,all |
| 01:18.04 | PBXtech | Atlanta has the best strippers |
| 01:19.24 | timecop | hm same fucking shit |
| 01:19.29 | timecop | so its not hte codec fault probably |
| 01:19.31 | timecop | god damn. |
| 01:19.40 | tengulre | I want use DELPHI develop client application. |
| 01:19.56 | tengulre | Agent Application. |
| 01:21.13 | tainted- | anyone have a good dialplan for receiving fax, converting to pdf, and e-mailing? |
| 01:21.27 | timecop | any way to find out what codec H323 channel is using? |
| 01:21.37 | timecop | i forced 723.1 in netmeeting, how can I confirm this? |
| 01:22.06 | PBXtech | you can make a dialplan to send variables to an external script to do the pdf and emailing |
| 01:22.13 | PBXtech | thats what i do |
| 01:22.24 | tengulre | tif2pdf? |
| 01:22.33 | tainted- | what format are faxes received in |
| 01:22.35 | PBXtech | something like that |
| 01:22.39 | PBXtech | tif |
| 01:22.43 | tengulre | tiff |
| 01:22.46 | *** part/#asterisk opus_ (opus@dahphish.org) |
| 01:22.49 | tengulre | yes :) |
| 01:23.32 | PBXtech | why are you so bent on 723? nothing much uses it |
| 01:24.35 | *** join/#asterisk jmav (~jmav@200.84.204.113) |
| 01:24.38 | timecop | because im trying to see if the fucking chinese voip provider that h323 calling fails to is bent on using that. |
| 01:24.41 | tainted- | PBXtech where are the faxes stored |
| 01:24.46 | timecop | because it doenst fuckign work |
| 01:24.50 | timecop | but netmeeting calls work. |
| 01:24.50 | tainted- | after a rxfax() |
| 01:24.59 | timecop | so something's fucked, calling to the chinks says cannot find compatible codecs. |
| 01:25.03 | jmav | Hello |
| 01:25.06 | timecop | even though I called them and they assure me they have 711u working. |
| 01:25.26 | *** part/#asterisk ropeguru (~ropeguru@fw.ropeguru.com) |
| 01:25.42 | PBXtech | well that 729 hack works well |
| 01:25.53 | timecop | well, i alreadyp aid for 729 |
| 01:25.54 | timecop | so I dont care |
| 01:25.55 | Sedorox | ~seen sleep |
| 01:25.57 | jbot | sleep <~dustin@c-24-16-13-109.client.comcast.net> was last seen on IRC in channel #tacobeam, 232d 18h 48m 37s ago, saying: 'haha'. |
| 01:25.58 | Sedorox | ~seen slepp |
| 01:25.59 | jbot | slepp <~slepp@S01060040f48412ad.ed.shawcable.net> was last seen on IRC in channel #asterisk, 2d 16h 45m 25s ago, saying: 'you did compile make opt as the docs say? i'm guessing so.'. |
| 01:27.17 | jmav | i have a question .... why if i make a sip call to my zaptel card works perfect and the audio its great... when i try to talk to another sip (2 sips toguether) the audio its cut (really bad connection .... i am doing something wrong in the configuration ? |
| 01:28.23 | Sedorox | bandwidth? |
| 01:28.41 | jmav | 384k/128k |
| 01:28.42 | timecop | i can place 2 concurrent 729 calls over a single 64k isdn channel |
| 01:28.59 | timecop | jmav: if youre using ulaw, that might be pushing it |
| 01:29.02 | timecop | use gsm or something lower. |
| 01:29.06 | Sedorox | yea... |
| 01:29.10 | Sedorox | especially on the upload |
| 01:29.33 | jmav | ohhh ok thx a lot where i can find the 729 ? |
| 01:29.35 | *** join/#asterisk TUplink (~Tommy@68-232-92-239.chvlva.adelphia.net) |
| 01:29.36 | timecop | you dont |
| 01:29.38 | timecop | you buy it |
| 01:29.38 | shepherd | i think you can get it down to 9kb/s with g729 |
| 01:29.39 | timecop | or |
| 01:29.41 | timecop | you use gsm |
| 01:29.42 | timecop | or ilbc |
| 01:29.48 | TUplink | how do i get jsut the phone number from callerid and not the name |
| 01:29.49 | timecop | shepherd: wat? |
| 01:29.57 | shepherd | maybe not |
| 01:29.57 | timecop | shepherd: i just said, I can do 2 729 calls over 64k isdn |
| 01:29.57 | shepherd | heh |
| 01:30.02 | timecop | thats like 7k/sec |
| 01:30.09 | timecop | 729 is about 24kbps/channel |
| 01:30.29 | TUplink | how do i get jsut the phone number from callerid and not the name? |
| 01:30.46 | jmav | if i Buy it. its easy to install ?? |
| 01:30.55 | JunK-Y | TUplink: CALLERIDNUM? |
| 01:31.03 | TUplink | ok... thx |
| 01:31.50 | timecop | jmav: www.digium.com |
| 01:31.53 | timecop | jmav: its $10/channel |
| 01:31.56 | timecop | yes its easy to install |
| 01:31.58 | PBXtech | what is gsm with overhead? |
| 01:32.02 | timecop | you just copy the .so to asterisk libs dir |
| 01:32.19 | timecop | PBXtech: 6k/sec or so |
| 01:32.34 | PBXtech | with overhead? cant be right |
| 01:32.37 | timecop | all i know is I cant do 2 gsm channels over isdn |
| 01:32.40 | timecop | but I can do one |
| 01:32.48 | timecop | so its < 7k/sec |
| 01:33.20 | shepherd | http://www.convergence.com.pk/iax2/trunked.html |
| 01:33.34 | shepherd | 34.8kb/s |
| 01:33.50 | shepherd | i think |
| 01:34.11 | Weezey | I hate sites that look like balls in firefox |
| 01:34.32 | timecop | nice nice |
| 01:34.54 | timecop | wow with trunked iax i can do 4 channels of 729 over single isdn channel? |
| 01:34.56 | timecop | wiN! |
| 01:35.07 | jmav | Thx a lot Timecop, Sedorox |
| 01:35.28 | shepherd | that's not including ip overhead |
| 01:35.35 | Sedorox | mm ok |
| 01:35.37 | shepherd | nm.. yes it is |
| 01:35.40 | shepherd | HAHA |
| 01:35.45 | timecop | thats fucking nice |
| 01:36.03 | timecop | but wait |
| 01:36.06 | timecop | 723 is even lower |
| 01:36.10 | timecop | i should jsut use that, haah. |
| 01:36.22 | timecop | 8 channels of 723 |
| 01:36.35 | timecop | hm |
| 01:36.36 | timecop | no |
| 01:36.37 | timecop | that wont work. |
| 01:38.48 | shepherd | munchies huh? |
| 01:39.35 | Sedorox | eh we have pizza.. |
| 01:41.22 | timecop | hmm |
| 01:41.44 | timecop | i guess I'll try a 723 channel over isdn later and see if that works |
| 01:41.55 | timecop | cant find a way to check status of h323 channel and its codec |
| 01:55.47 | *** join/#asterisk ToyMan (~konversat@user-12lcqur.cable.mindspring.com) |
| 02:01.24 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 02:01.24 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 02:01.40 | *** part/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 02:01.48 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 02:01.48 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 02:05.36 | timecop | hm |
| 02:05.37 | timecop | nice and dead |
| 02:05.43 | timecop | good to see that asterisk is a 100% us-centric product |
| 02:07.56 | Qwell | So, when did it lose support for things like E1, and other non-us things? |
| 02:08.08 | *** join/#asterisk DannyF (~dannyf@h22n7c1o848.bredband.skanova.com) |
| 02:08.52 | Sedorox | <PROTECTED> |
| 02:14.24 | Dovid | anyone have a problem with the smc baricade router ? |
| 02:16.46 | Dovid | anyone ? |
| 02:16.57 | JerJer | i have never even heard of it |
| 02:17.03 | JerJer | so i suspect others haven't either |
| 02:17.04 | Dovid | ~seen [SuPrSluG] |
| 02:17.06 | jbot | Dovid: i haven't seen '[suprslug]' |
| 02:17.16 | Qwell | Its just a generic broadband router, isn't it? |
| 02:17.21 | Dovid | yes |
| 02:17.29 | Dovid | it wont forward 10,000 - 20,000 |
| 02:17.34 | Dovid | ~seen suprslug |
| 02:17.35 | jbot | suprslug is currently on #asterisk (13h 8m 10s). Has said a total of 6 messages. Is idling for 12h 24m 47s |
| 02:18.28 | Qwell | hey JerJer, got a second? |
| 02:18.41 | Qwell | for a pm, that is. Have an interesting(maybe?) idea for you |
| 02:19.38 | *** join/#asterisk tengulre (~tengulre@61.185.238.166) |
| 02:19.50 | JerJer | i have lots of seconds |
| 02:22.29 | *** join/#asterisk Entegrity (~Entegrity@c-65-96-119-254.hsd1.ma.comcast.net) |
| 02:23.41 | Entegrity | Do I need to purchase softphone from Vonage or will asterisk work without it? |
| 02:27.57 | *** join/#asterisk mog_home (~mog_home@146.229.176.225) |
| 02:30.23 | bjohnson | JerJer: any firsts? |
| 02:30.39 | bjohnson | Entegrity: you need to give up on vonage |
| 02:30.52 | *** join/#asterisk nev4 (~nevspam@pool-70-21-92-152.res.east.verizon.net) |
| 02:31.02 | nev4 | gr33tz |
| 02:31.05 | nev4 | :) |
| 02:31.23 | nev4 | anyone available to give me a quick hand with a SIP 302 issue? |
| 02:32.02 | Entegrity | give up? |
| 02:32.13 | nev4 | sorry, did I miss some protocol? should I send flowers first? |
| 02:32.26 | bjohnson | Entegrity: you need to buy the softphone account in addition to the regular account. |
| 02:32.39 | *** join/#asterisk evo4wrx (~sdfg@dsl-202-72-150-77.wa.westnet.com.au) |
| 02:32.43 | Entegrity | Unless I go with an FXO... |
| 02:32.44 | bjohnson | minutes from the two accounts are not pooled .. they are billed as separate entitiies |
| 02:32.57 | Entegrity | hmm I have router with an FXO... but I dunno if it does SIP. |
| 02:32.59 | Qwell | Entegrity: which is a silly method |
| 02:33.05 | evo4wrx | has anyone ever come across the crackle with quad span e1's with the IRQ |
| 02:33.10 | bjohnson | there are hundreds of voip companies that will likely give you a better price than vonage |
| 02:33.19 | Entegrity | Hmm |
| 02:33.27 | bjohnson | Entegrity: you need to give up on vonage |
| 02:33.32 | Entegrity | :| |
| 02:33.52 | nev4 | yeah, I'll second that, BroadVoice is better in just about every way |
| 02:34.01 | Qwell | broadvoice? Not really. heh |
| 02:34.18 | *** join/#asterisk tessier (~treed@203.210.209.79) |
| 02:34.19 | nev4 | at least they open their service up |
| 02:34.45 | nev4 | prepaid is a little on the unprofessional side, okay for b2C, but not for most companies |
| 02:35.01 | evo4wrx | can someone help me? |
| 02:35.02 | at561 | too bad so many places require $20 minimums |
| 02:35.21 | nev4 | $20 is nothing, how about $10,000 just to offer you DID's |
| 02:35.50 | at561 | being a starving student i can't afford to leave $20 of unspent minutes in some account |
| 02:35.52 | nev4 | or I should say to be eligible for origination services |
| 02:36.01 | nev4 | right |
| 02:36.34 | at561 | voip is my gateway to a one dollar a month phone line |
| 02:36.35 | nev4 | sorry t obe pushy, but is anyone familiar with asterisk's interaction with SIP 302 redirects? |
| 02:37.42 | bjohnson | at561: I thought iax.cc ws $10 minimum deposit |
| 02:37.44 | mgth | $20 is a sunk cost |
| 02:37.50 | bjohnson | well .. $12 something |
| 02:38.03 | bjohnson | what is a sunk cost? |
| 02:39.06 | ManxPower | nev4, What specific issue do you have? |
| 02:39.11 | mgth | http://en.wikipedia.org/wiki/Sunk_cost |
| 02:39.32 | nev4 | I am working with a termination provider who uses a 302 redirect |
| 02:39.47 | nev4 | calls don't go through when I send them to that IP that redirects, I get a SIP 404 |
| 02:39.58 | tengulre | morning,all! |
| 02:40.02 | nev4 | it tries SIP/theextension@onethierotherIPS |
| 02:40.13 | nev4 | sorry, @ oneoftheirotherIPs |
| 02:40.19 | nev4 | norin' |
| 02:40.28 | *** join/#asterisk marcus5 (~marcus@pompeii.outer.org) |
| 02:41.20 | marcus5 | will the tdm400p work in a 3.3v pci slot? |
| 02:41.32 | Qwell | 3.3 or 5 I thought |
| 02:41.38 | Qwell | It says it on the page... |
| 02:41.49 | evo4wrx | have fun with the crackle |
| 02:42.02 | marcus5 | it doesnt explicitly say |
| 02:42.06 | marcus5 | crackle? |
| 02:42.21 | Qwell | well, it said it somewhere |
| 02:42.32 | Qwell | ahh, heh |
| 02:42.38 | Qwell | on the card they shipped with it |
| 02:42.43 | nev4 | any ideas Manx? |
| 02:42.48 | marcus5 | it looks like the card has two notches in it |
| 02:42.51 | evo4wrx | ooo |
| 02:42.52 | evo4wrx | nah |
| 02:42.54 | marcus5 | so i'm guessing its universal |
| 02:42.56 | evo4wrx | that one doesnt crackle |
| 02:43.04 | evo4wrx | te410 does |
| 02:43.09 | Qwell | "3. Insert the TDM400p into a 3.3- or 5-volt PCI slot (PCI 2.2 or greater required" |
| 02:43.16 | marcus5 | really |
| 02:43.18 | Qwell | Looks like they forgot the end brace |
| 02:43.23 | marcus5 | ok cool |
| 02:44.35 | MarkS__ | does asterisk have a counter variable of like the # caller or whatever? |
| 02:45.34 | bjohnson | you could just add one to a global variable |
| 02:45.38 | bjohnson | or use setgroup |
| 02:46.26 | tengulre | hi, how can i communicate between Server and Client? In LAN? |
| 02:46.54 | *** join/#asterisk doughecka_ (~Doug@doughecka.user) |
| 02:47.01 | bjohnson | sip or iax are what the cools kids are using these days |
| 02:47.05 | *** join/#asterisk montoya (montoya@200.195.90.185) |
| 02:47.27 | Qwell | bjohnson: I prefer qgp |
| 02:47.32 | doughecka_ | sip != cool :P |
| 02:48.06 | MarkS__ | how do i do that bjhonson? |
| 02:48.15 | MarkS__ | *bjohnson - |
| 02:48.17 | bjohnson | MarkS__: read |
| 02:48.19 | tengulre | bjohnson,if In Lan? |
| 02:48.20 | bjohnson | ~docs |
| 02:48.21 | 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 |
| 02:48.44 | bjohnson | MarkS__: read about setvar(), read about variables, and read about setgroup() |
| 02:49.01 | bjohnson | tengulre: in, out, wherever |
| 02:49.19 | *** join/#asterisk iq|laptop (~iq@70-59-160-50.omah.qwest.net) |
| 02:49.41 | tengulre | bjohnson, which site provide more messages? |
| 02:50.48 | bjohnson | tengulre: not really |
| 02:51.33 | tengulre | bjohnson,:) sorry,I'm beginner! |
| 02:53.35 | *** join/#asterisk Trepalium (~chadk@wnpgmb02dc1-57-202.dynamic.mts.net) |
| 02:54.40 | tengulre | bjohnson,I want use DELPHI develop client application? but i dont know how communicat,as so softphone . |
| 02:56.58 | nev4 | anyone available to give me a quick hand with a SIP 302 redirect issue? |
| 02:59.09 | tengulre | bjohnson, are you here? |
| 03:01.20 | PTG1234 | tengulre: you need a delphi component for a softphone? |
| 03:03.10 | *** join/#asterisk slePP (~slepp@S01060040f48412ad.ed.shawcable.net) |
| 03:04.33 | *** join/#asterisk reph (~r@user-24-236-86-244.knology.net) |
| 03:06.14 | tengulre | PTG1234,tks, which component? do you know? |
| 03:08.06 | PTG1234 | is that what you need? |
| 03:08.45 | file[laptop] | PTG1234: guess what clears tomorrow |
| 03:09.17 | PTG1234 | hah |
| 03:09.22 | PTG1234 | its about f'ing time :) |
| 03:09.36 | file[laptop] | hehe |
| 03:09.40 | *** join/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 03:09.49 | PTG1234 | i better run some burn in tests on those suckers tonight, make sure they are in tip top shape :) |
| 03:10.26 | file[laptop] | haha |
| 03:10.35 | PTG1234 | i'll do a makie world on em :) |
| 03:10.44 | PTG1234 | did you get your pda yet? |
| 03:10.55 | file[laptop] | it'll be delivered tomorrow |
| 03:11.25 | file[laptop] | UPS Has been holding it for the last two days where it was picked up as they have a daily flight from there to here... so they're just going to stuff it on that I guess |
| 03:13.55 | PTG1234 | ah cool |
| 03:14.01 | PTG1234 | i have learned all sorts of pday tricks :) |
| 03:14.09 | file[laptop] | ooh anything extremely fun? |
| 03:15.10 | PTG1234 | heh not sure exactly :) |
| 03:15.15 | PTG1234 | mine is now connected to the inet 24/7 :) |
| 03:15.23 | PTG1234 | notifies my of new mail, aim, msn, blah blah |
| 03:15.25 | PTG1234 | its cool :) |
| 03:15.27 | file[laptop] | I had planned on that anyway |
| 03:15.42 | PTG1234 | yah but its connected and can still receive calls |
| 03:15.50 | PTG1234 | thanks to a registry hack |
| 03:15.52 | file[laptop] | haha |
| 03:15.58 | file[laptop] | yours unlocked? |
| 03:16.02 | file[laptop] | or did you get CDMA? |
| 03:16.09 | PTG1234 | um nah, it can be easily though you just need the MSL |
| 03:16.16 | PTG1234 | its CDMA but it can be unlocked to do other CDMA networks |
| 03:16.21 | file[laptop] | cool |
| 03:16.33 | PTG1234 | but yoiu know we get EVDO soon, sprint is suppose to have the largest EVDO network, so why would i want to unlock it and go elsewhere :) |
| 03:16.51 | file[laptop] | because they're probably lying and it won't turn out that good? :) |
| 03:17.05 | PTG1234 | i don';t know :) |
| 03:17.13 | PTG1234 | people in kansas city say it is :) |
| 03:17.19 | file[laptop] | all lies! |
| 03:17.20 | PTG1234 | which is where it was just launched |
| 03:17.21 | PTG1234 | hah |
| 03:17.23 | *** join/#asterisk [shodan] (~shodan@216.113.99.170) |
| 03:17.37 | [shodan] | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=67290&item=5767706862&rd=1&ssPageName=WDVW is this any good ? |
| 03:18.36 | PTG1234 | get the soyo |
| 03:19.12 | WilliamK | shodan, all I can say is beware of the counterfit cisco look alikes from china |
| 03:19.18 | PTG1234 | 4 ports and like $50 |
| 03:19.38 | *** join/#asterisk cc (~cc@byte.fedora) |
| 03:20.16 | PTG1234 | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=1503&item=5763669707&rd=1&ssPageName=WD1V |
| 03:20.17 | PTG1234 | $27 |
| 03:20.20 | PTG1234 | shit i am buying one now :) |
| 03:21.15 | file[laptop] | resist! |
| 03:21.34 | PTG1234 | Mmmmmmmmmmmm 4 fxs ports |
| 03:21.37 | [shodan] | 28$ for 4 fxs |
| 03:21.40 | [shodan] | holyshit |
| 03:21.43 | PTG1234 | yah |
| 03:21.47 | PTG1234 | you owe me one :) |
| 03:21.51 | [shodan] | hehe |
| 03:21.55 | file[laptop] | or two. |
| 03:22.24 | [shodan] | I was like , there's no I'm spending more than 25$ per fxs , and this is 4 !!! |
| 03:23.01 | PTG1234 | yah t.38 support too :) |
| 03:23.05 | PTG1234 | if only asterisk supported it |
| 03:23.11 | PTG1234 | hey file you gonna try and program on that pda? :) |
| 03:23.27 | file[laptop] | yes, and I'm also going to fly around town on a broom |
| 03:23.36 | remmo | lol |
| 03:23.44 | file[laptop] | care to join me? :P |
| 03:23.49 | PTG1234 | sure :) |
| 03:24.01 | PTG1234 | the programming interface on that makes me almost want to use visual basic :) |
| 03:24.11 | file[laptop] | well, you can |
| 03:24.14 | PTG1234 | a simple hello world app is like 200 lines |
| 03:24.25 | remmo | only if we could get gecko on a pda |
| 03:24.25 | file[laptop] | use an MFC Appwizard |
| 03:24.27 | file[laptop] | much quicker... |
| 03:24.31 | file[laptop] | easier |
| 03:24.32 | WilliamK | anyone use the soyo's yet? |
| 03:24.42 | PTG1234 | its make the interfaces that seem to take forever |
| 03:24.46 | PTG1234 | file: you know vc++? |
| 03:25.00 | file[laptop] | PTG1234: yes, which is why I suggest using the above because it's uber-simple |
| 03:25.10 | file[laptop] | point and click to design your interface, then do your controls |
| 03:25.51 | PTG1234 | um huh |
| 03:26.25 | file[laptop] | you need the mobile vc++ btw |
| 03:26.29 | PTG1234 | yah got it |
| 03:26.32 | PTG1234 | vc++ embedded |
| 03:26.45 | file[laptop] | but if it's there, it should be called an MFC Appwizard... hopefully it has the capacity for those |
| 03:26.51 | PTG1234 | i pick file new |
| 03:26.55 | PTG1234 | and then what? |
| 03:27.06 | file[laptop] | to be honest I couldn't get the thing installed on my laptop |
| 03:27.11 | PTG1234 | ok |
| 03:27.14 | PTG1234 | got the app wizard |
| 03:27.19 | file[laptop] | choose dialog |
| 03:27.21 | PTG1234 | you coldn't get embedded installed? |
| 03:27.31 | file[laptop] | correct, Windows errored the installer out |
| 03:27.42 | Qwell | file[laptop]: Thats unexpected? |
| 03:27.54 | PTG1234 | ok it created something |
| 03:28.02 | PTG1234 | now how do i make things on it :) |
| 03:28.09 | Qwell | PTG1234: click away at the controls, heh |
| 03:28.20 | PTG1234 | where is the wizzzzy wig |
| 03:28.35 | file[laptop] | go into the Resources... Dialogs... choose the main dialog, and it should pop up |
| 03:28.45 | file[laptop] | you can move controls around... do whatever, add new ones |
| 03:29.03 | file[laptop] | double click to create/edit the function that is executed when they are clicked |
| 03:29.14 | file[laptop] | hit Ctrl+W and then member variables to assocate variables with edit boxes and junk |
| 03:29.25 | file[laptop] | Ctrl+R to add icons/bitmaps to the resources, or create your own |
| 03:29.48 | file[laptop] | I have a basic manual here if you'd like it |
| 03:30.01 | file[laptop] | it's the one my school uses that I ignore |
| 03:30.06 | dizzydiffi | heelo |
| 03:30.11 | file[laptop] | hi |
| 03:30.19 | dizzydiffi | has anyone dabbled with open h323 |
| 03:30.52 | dizzydiffi | anyone |
| 03:31.06 | Nugget | I think that's a "no" |
| 03:31.13 | dizzydiffi | gosh |
| 03:31.27 | Nugget | all I know about h323 is that everyone who mentions it is asking how to make it work. :) |
| 03:31.34 | Nugget | doesn't bode well |
| 03:31.38 | dizzydiffi | i got it to work |
| 03:31.42 | dizzydiffi | well half way |
| 03:31.44 | Trepalium | And no one ever gives any answers. |
| 03:32.05 | jakepdev | h323 implementaion is one of the better kept secrets of * |
| 03:32.05 | dizzydiffi | i just cant call out but i can recieve calls |
| 03:32.13 | dizzydiffi | great |
| 03:32.32 | jakepdev | did you follow the readme to the T? |
| 03:32.36 | tzafrir_laptop | hi, someone asked me if Asterisk supports h223 (not a typo) |
| 03:32.54 | JerJer | define supports |
| 03:33.01 | dizzydiffi | exactly |
| 03:33.05 | dizzydiffi | its crazy |
| 03:33.10 | tzafrir_laptop | Supposed to be part of H245 rev. 10 or something |
| 03:34.01 | tzafrir_laptop | dizzydiffi, I know I can build openh323. never tried to actually use it |
| 03:34.19 | file[laptop] | did we lose PTG1234? |
| 03:34.30 | PTG1234 | damn cell phone rang |
| 03:34.31 | dizzydiffi | you know what after like 1 week i got it to build and i got a sip phone to dial |
| 03:34.31 | PTG1234 | one sec |
| 03:34.35 | dizzydiffi | the h323 |
| 03:34.38 | file[laptop] | k, as long as we didn't lose you |
| 03:34.39 | dizzydiffi | and it worked |
| 03:35.06 | dizzydiffi | but i just cant call from h323 |
| 03:35.13 | file[laptop] | oh cool my pocket pc phone is in the local depot |
| 03:35.21 | slePP | 29 more posts to the pastebin and it hits 10,000 :> |
| 03:35.29 | *** join/#asterisk sergiomiguelrp (sergiomigu@200.84.218.222) |
| 03:35.38 | file[laptop] | actually it was there yesterday |
| 03:35.41 | tzafrir_laptop | JerJer, a google search did not bring out much about it |
| 03:35.58 | file[laptop] | technically yesterday cause today is the 21st |
| 03:36.50 | slePP | 100% nuts |
| 03:36.58 | file[laptop] | what kind? |
| 03:37.03 | slePP | almond |
| 03:37.08 | JerJer | niggertoe |
| 03:37.23 | sergiomiguelrp | Hi I'm a NEWBIE, who may i talk to? |
| 03:37.25 | JerJer | damnit i wanted the crack one |
| 03:37.27 | JerJer | :( |
| 03:37.54 | file[laptop] | yay crack |
| 03:38.09 | tzafrir_laptop | JerJer, so is it something implemented in asterisk/h323 and/or openh323? |
| 03:38.23 | dizzydiffi | hey tzafrir_laptop |
| 03:38.24 | jakepdev | sergiomiguelrp: just ask what you want to ask |
| 03:38.24 | iq|laptop | sergiomiguelrp, everyone |
| 03:38.33 | dizzydiffi | what you asing about openh323 |
| 03:38.56 | tzafrir_laptop | if it supports something called h223 |
| 03:39.04 | file[laptop] | poor slePPyboy |
| 03:39.19 | slePP|AX | oh, yeh. no .'s in the nick |
| 03:39.31 | dizzydiffi | well i got openh323 to work on asterisk |
| 03:40.12 | iq|laptop | is g723.1 free to use? |
| 03:40.12 | file[laptop] | iq|laptop: no. |
| 03:40.12 | dizzydiffi | as least i can recieve calls from SIP |
| 03:40.12 | file[laptop] | highly licensed... expensive |
| 03:40.18 | iq|laptop | file[laptop], more than g729? |
| 03:40.18 | file[laptop] | expensive on 'da wallet, and 'da CPU |
| 03:40.24 | iq|laptop | oops |
| 03:40.31 | file[laptop] | well you can't get it for asterisk |
| 03:40.41 | file[laptop] | unless you wanna do it yourself. |
| 03:40.44 | file[laptop] | atleast legally |
| 03:41.00 | iq|laptop | hmmm |
| 03:42.45 | dizzydiffi | does it actually work |
| 03:42.55 | dizzydiffi | h323 in asterisk i mean |
| 03:43.02 | dizzydiffi | if not body seems to know how to make it work |
| 03:43.55 | tzafrir_laptop | hmmm, used the wrong search: "H.223" in google suddenly gives results :-) |
| 03:44.04 | reph | I wanna make a PBXbox |
| 03:44.16 | reph | so I can recieve calls and play halo |
| 03:44.55 | jakepdev | reph - I believe there is documentation on the wiki about that |
| 03:45.41 | jakepdev | http://nlug.org/mail/nlug%5F%5F2003_12/0094.html |
| 03:45.58 | tzafrir_laptop | dizzydiffi, how exactly? What * version? |
| 03:46.04 | *** join/#asterisk soundguy (~soundguy@zeus.blendtek.com.au) |
| 03:46.25 | dizzydiffi | the latest version |
| 03:46.28 | dizzydiffi | of asterisk |
| 03:46.31 | dizzydiffi | 1.0.7 |
| 03:46.42 | jakepdev | does it work in stable? |
| 03:46.48 | jakepdev | i think you need head for that |
| 03:46.55 | *** join/#asterisk Tond (Tond@Toronto-HSE-ppp3646966.sympatico.ca) |
| 03:47.06 | dizzydiffi | yea it works |
| 03:47.17 | soundguy | How do you show current IAX registrations in the asterisk console? |
| 03:47.27 | tzafrir_laptop | chan_h323 seems to be aiming for HEAD. The Debian package has a certain version of chan_oh323 |
| 03:47.38 | dizzydiffi | i use oh323 |
| 03:47.51 | Qwell | soundguy: iax2 show registry |
| 03:48.10 | tzafrir_laptop | We have 0.6.6pre3 packages, IIRC |
| 03:48.13 | soundguy | *CLI> iax show registry |
| 03:48.13 | soundguy | No such command 'iax' (type 'help' for help) |
| 03:48.15 | Tond | Hi.. Is there any codec except for G711 I can use to establish connection between a Cisco router and my * box, excluding G729? |
| 03:48.21 | Qwell | soundguy: read what I wrote |
| 03:48.23 | jakepdev | iax2 not iax |
| 03:48.28 | soundguy | ahh |
| 03:49.09 | soundguy | this is weird, I am getting 'Call rejected by 220.233.127.6: No authority found', but I am registered?? |
| 03:49.23 | jakepdev | Tond - that all depends what codecs your Cisco router accepts |
| 03:49.40 | dizzydiffi | soundguy wat r u using |
| 03:49.49 | soundguy | as in? |
| 03:49.50 | jakepdev | Tond: http://64.233.161.104/search?q=cache:sInb-1a9DtsJ:www.voip-info.org/wiki-Asterisk%2Bcodecs+asterisk+codecs&hl=en |
| 03:49.59 | Tond | In other words can I use GSM codec to connect the call between my Cisco 3600 or 5350? Since Cisco supports GSM-EFR and GSM-FR, but * says it suports GSM. Are they the same? |
| 03:50.06 | Tond | thanks... |
| 03:51.15 | *** join/#asterisk Hackett (~chatzilla@cuscon2673.tstt.net.tt) |
| 03:52.37 | jakepdev | soundguy: try an iax2 debug... it my give you a better clue as to what is happening |
| 03:53.02 | PTG1234 | ok file still around |
| 03:53.14 | Hackett | I am look for a good iax provider besides www.iax.cc or voicepulse |
| 03:53.22 | jakepdev | nufone? |
| 03:53.26 | file[laptop] | PTG1234: I appear to be. |
| 03:53.27 | jakepdev | livevoip |
| 03:53.37 | jakepdev | voipjet |
| 03:53.44 | PTG1234 | ok got my dialog up |
| 03:53.48 | PTG1234 | so i um.. hmm :) |
| 03:53.55 | file[laptop] | experiment! |
| 03:54.04 | PTG1234 | how do i show a message |
| 03:54.11 | PTG1234 | like showmessage("blah"); |
| 03:54.16 | PTG1234 | just pop up a little box with a message on it |
| 03:54.23 | tzafrir_laptop | hmmm... http://www.voip-info.org/wiki-Asterisk+H324M seems to suggest that h322 is not implemented yet |
| 03:54.40 | file[laptop] | AfxMessageBox("You sir, are an idiot"); |
| 03:54.52 | PTG1234 | actually how do i change properties of an object, like its same? |
| 03:54.54 | PTG1234 | er its name |
| 03:55.00 | file[laptop] | right click it |
| 03:55.05 | file[laptop] | and go into Properties |
| 03:55.17 | PTG1234 | i did that |
| 03:55.19 | PTG1234 | nothing pops up |
| 03:55.27 | file[laptop] | O.o |
| 03:55.56 | file[laptop] | a Properties window should pop up... |
| 03:55.59 | PTG1234 | what hehell :) |
| 03:56.01 | file[laptop] | try selecting the control and then typing |
| 03:57.21 | PTG1234 | that worked |
| 03:57.33 | PTG1234 | ok so i have an edit box |
| 03:57.36 | PTG1234 | how do i change the text in the box |
| 03:57.53 | PTG1234 | no code completion in vc++ how quaint :) |
| 03:58.11 | file[laptop] | it's made for grade 12 students in high school, so you'll take to it fine *G* |
| 03:58.15 | PTG1234 | is it an ebook, so i can use it on my pda.. yay |
| 03:58.21 | file[laptop] | it's a huge word document |
| 03:58.21 | PTG1234 | i'm a delphi man :) |
| 03:58.23 | file[laptop] | well not huge |
| 03:58.24 | PTG1234 | this vc++ is shitty |
| 03:58.30 | file[laptop] | but it's 6.3MB |
| 03:58.53 | PTG1234 | just tell me that last question |
| 03:58.55 | PTG1234 | and i am good :) |
| 03:58.56 | file[laptop] | please hold while it uploads |
| 03:59.01 | file[laptop] | this'll tell you EVERYTHING |
| 03:59.07 | file[laptop] | you need to know. |
| 03:59.19 | PTG1234 | apparently vc++ doesn't like focus follows mouse |
| 04:00.51 | file[laptop] | http://file-radio.com/manual.doc |
| 04:01.10 | *** join/#asterisk Syncros (~sysop@noc.routermonkey.net) |
| 04:01.28 | Sedorox | anyone here of Asterisk locking up when loading... spefically loading musiconhold? |
| 04:02.44 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 04:05.27 | sergiomiguelrp | mosichold |
| 04:05.31 | sergiomiguelrp | musichold |
| 04:05.39 | sergiomiguelrp | musiconhold |
| 04:06.06 | PTG1234 | i like the little phone emulators :) |
| 04:06.20 | file[laptop] | PTG1234: hehe |
| 04:06.44 | PTG1234 | its the phone apps that give me a headache, b/c they are not dialog based |
| 04:08.56 | [shodan] | anyone bought a N400S ? is it as good as the spec sheet says ? |
| 04:10.33 | WilliamK | anyone know of a good USB based phone? |
| 04:12.43 | *** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode) |
| 04:13.34 | PTG1234 | come on file, asterisk on ppc :) |
| 04:13.40 | PTG1234 | want a good challenge |
| 04:14.29 | WilliamK | thought it was already done |
| 04:14.46 | file[laptop] | PTG1234: haha ... no. |
| 04:15.04 | file[laptop] | although it would help if it got here, hint hint UPS |
| 04:15.47 | JerJer | PTG1234: why is asterisk ppc a challenge? |
| 04:16.18 | JerJer | zaptel, maybe |
| 04:16.21 | JerJer | but asterisk doesn't care |
| 04:16.24 | JerJer | at all |
| 04:16.37 | file[laptop] | he means pocket pc |
| 04:16.57 | PTG1234 | well for starters its windows :) |
| 04:17.00 | PTG1234 | and a scaled down windows |
| 04:17.52 | JerJer | don't need pocket pc |
| 04:17.59 | JerJer | just run linux on the target cpu |
| 04:18.16 | PTG1234 | where would be the fun in that :) |
| 04:18.37 | PTG1234 | i have a sl-5500 with linux on it, not quite as fun |
| 04:19.48 | *** join/#asterisk cc (~cc@byte.fedora) |
| 04:21.52 | PTG1234 | think its overkill to have my pda check my email every 2 minutes |
| 04:21.56 | [shodan] | !!!! the N400S can do both fxs and fxo ! |
| 04:22.15 | WilliamK | shodan, you reading the user manual too? |
| 04:22.21 | file[laptop] | PTG1234: nah |
| 04:22.22 | PTG1234 | shodan: not exactly |
| 04:22.34 | file[laptop] | PTG1234: but if you had something to push new e-mail to your PDA instantly, that would be better |
| 04:22.59 | PTG1234 | file: you can |
| 04:23.06 | PTG1234 | through sms message, it can trigger to retrieve your email |
| 04:23.12 | PTG1234 | although haven't found out how yet |
| 04:23.15 | file[laptop] | meh not good enough |
| 04:23.30 | PTG1234 | it saids and sms message that basically tells your pda get my mail |
| 04:24.01 | PTG1234 | thats how the blackberries work |
| 04:24.27 | JerJer | we tested a wifi blackberry with a SIP phone |
| 04:25.02 | [shodan] | mm , well I'm not sure now , it said "FXS/FXO interface" , but it's not ? |
| 04:25.05 | PTG1234 | the only problem with sip phones is none support g729 on a pda |
| 04:25.07 | JerJer | we got java exceptions all over the place :( |
| 04:25.22 | PTG1234 | but they claim once EVDO comes out, you can just use sip on these things and never need to use them as a cell phone :) |
| 04:25.30 | nestAr | argh |
| 04:26.12 | nestAr | I can't get SetGroup / CheckGroup to work with CVS-HEAD |
| 04:29.55 | Tond | JerJer> That si really interesting, cause my new job is going to be with RIM supporting the new WiFi BlackBerry |
| 04:30.20 | Tond | JerJer> So you were not able to establish calls then, right? |
| 04:35.13 | *** join/#asterisk AgiNamu (~zzzs@216.230.151.230) |
| 04:35.16 | AgiNamu | hey all |
| 04:35.18 | nestAr | hi |
| 04:35.28 | AgiNamu | I'm trying to profile asterisk using google-perftools |
| 04:35.35 | AgiNamu | I added -lprofiler and compiled |
| 04:35.36 | AgiNamu | now I get |
| 04:35.43 | AgiNamu | asterisk: error while loading shared libraries: libprofiler.so.0: cannot open shared object file: No such file or directory |
| 04:35.46 | *** join/#asterisk riksta (~rick@81-178-209-106.dsl.pipex.com) |
| 04:35.56 | AgiNamu | wtf? It's in /usr/local/lib, I tried putting it all over . set LD_LIBRARY_PATH |
| 04:36.51 | Tond | AgiNamu> what will that allow you to do? to profile * using google-perftools...? |
| 04:38.15 | nestAr | oops, i just rm -rf /usr/src/* |
| 04:38.56 | iq | nestAr, turn off playboy |
| 04:39.02 | Tond | nestAr> ur still lucky u didn't do rm -rf / |
| 04:39.08 | Tond | lol |
| 04:39.17 | Tond | seriously.. lol |
| 04:39.21 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 04:40.50 | nestAr | yeah |
| 04:40.52 | nestAr | i guess so |
| 04:40.58 | nestAr | i'm just tired and frustrated |
| 04:48.21 | *** join/#asterisk anachron (phr34k@ns1.ts-shells.com) |
| 04:48.24 | JerJer | Tond: we could make calls and when they worked they were solid |
| 04:48.31 | anachron | anyone familiar with static configuration of the IAXy? |
| 04:48.39 | JerJer | but lots of things made the SIP phone app crash |
| 04:48.40 | anachron | i can't get mine to respond to provisioning |
| 04:48.52 | Tond | Hrm... |
| 04:48.53 | JerJer | like call waiting |
| 04:49.10 | JerJer | then getting rejected by the proxy |
| 04:49.27 | JerJer | then other times the app would crash on incoming calls |
| 04:49.28 | Qwell | Which phones? |
| 04:49.34 | JerJer | RIM wifi |
| 04:49.38 | Tond | Well it is at it's early stages... long way to go |
| 04:49.41 | JerJer | yeah |
| 04:49.46 | JerJer | open source it |
| 04:49.58 | Tond | If it was up to me i would ahve.. :D |
| 04:50.08 | Tond | have |
| 04:50.55 | Tond | I am trying to find a way to connect Cisco routers to * with a codec other than G711 |
| 04:51.22 | Tond | is there a specific IOS that supports a codec that * can support as well? |
| 04:52.06 | PTG1234 | g729 |
| 04:52.30 | Tond | Well that is a code that needs to be licensed per channel |
| 04:53.13 | Tond | plus cisco routers support G729R and Br, and I beleive asteriisk's G729 is G729 AnnexA |
| 04:54.11 | *** join/#asterisk SplasPood (jwb@schizophrenia.paravolve.net) |
| 04:57.16 | ScythelX | has anyone had problems installing asterisk-addons with freebsd |
| 04:59.11 | *** join/#asterisk SplasPood (jwb@paravolve.net) |
| 05:00.22 | *** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:00.47 | *** join/#asterisk Nukemizer (~Nuke@66.237.85.58.ptr.us.xo.net) |
| 05:02.32 | Moc[Toronto] | Hail |
| 05:06.45 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 05:06.46 | Nukemizer | Is there a way to stop an Xlite softphone from getting a second ACD call while on an existing ACD call ? |
| 05:15.22 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 05:18.44 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 05:24.01 | *** join/#asterisk MrEntropy (~entropy@155.003.dsl.sa.iprimus.net.au) |
| 05:24.03 | MrEntropy | is h.261 an open video codec? |
| 05:28.48 | *** join/#asterisk ellvis (~ellvis@195.98.29.34) |
| 05:28.52 | ellvis | hi people |
| 05:29.01 | MrEntropy | allright, does anyone know of any open source codec implementations of either h.261 or h.263? |
| 05:32.21 | MrEntropy | what is considered the cosher way for two asterisks to communicate(forward calls to each other)? I imagine just putting each other in sip.conf is not the nicest thing to do, is it? |
| 05:34.28 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
| 05:34.38 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 05:35.10 | Qwell | MrEntropy: like iax |
| 05:35.19 | Qwell | with trunking and all |
| 05:35.23 | Qwell | likely* |
| 05:35.56 | MrEntropy | right, so iax is THE THING to use for two asterisks? |
| 05:36.01 | tengulre | Qwell,Cool! |
| 05:36.04 | MrEntropy | i'm being vague, i know |
| 05:36.13 | Qwell | well, it IS Inter-Asterisk eXchange |
| 05:36.25 | MrEntropy | oh, i see, how silly of me |
| 05:36.29 | drumkilla | indeed - in general, yes |
| 05:38.16 | tengulre | where have completed documents about IAX? |
| 05:40.26 | MrEntropy | i just found some at voip-info |
| 05:40.30 | tengulre | does the asterisk support ctServer? |
| 05:43.16 | tengulre | can the asterisk control agents in LAN? |
| 05:44.54 | *** join/#asterisk cjk (~cjk@80.92.75.232) |
| 05:45.16 | cjk | hi, is transcoding a cpu or a memory intensive options? |
| 05:47.23 | Moonwick | cpu |
| 05:54.19 | *** join/#asterisk _solstice_ (~solstice@dsl-cap-209-5-169-205-cgy.nucleus.com) |
| 05:55.58 | cjk | Moonwick: do you know how much is used for ilbc |
| 05:57.55 | PTG1234 | just reg'd andreasemail.com for my wife :) |
| 05:58.53 | Qwell | I like my (highly ironic) email domain... |
| 05:59.02 | PTG1234 | whats that? :) |
| 05:59.07 | Qwell | ntbox.com |
| 05:59.26 | PTG1234 | i don't get it :) |
| 05:59.31 | Qwell | Its running on a Linux box... |
| 05:59.36 | PTG1234 | ah i gotcha |
| 05:59.49 | PTG1234 | i spent 2 days getting this mail box online |
| 05:59.51 | Qwell | kinda like hotmail I guess |
| 05:59.55 | PTG1234 | now i remember why i wrote my own mail server :) |
| 05:59.58 | Qwell | heh |
| 06:00.04 | PTG1234 | now i got to make my mailserver iface with all thise lovely qmail crap |
| 06:00.07 | Qwell | Which mailserver did you use? |
| 06:00.19 | PTG1234 | qmail + tls |
| 06:00.31 | PTG1234 | vqadmin, qmailadmin, sqwebmail, authdaemon, vpopmail |
| 06:00.45 | Qwell | I like sqmail |
| 06:00.59 | PTG1234 | dovecot too |
| 06:01.04 | PTG1234 | yah its pretty decient |
| 06:01.07 | Qwell | yeah, imap is needed |
| 06:01.10 | PTG1234 | i want to develop my own replacements for all of that |
| 06:01.19 | Qwell | good luck, heh |
| 06:01.19 | PTG1234 | but i want a fully working system while i do it |
| 06:01.23 | PTG1234 | so i can replace one piece at a time |
| 06:02.11 | Qwell | god I love newsgroups |
| 06:02.21 | Qwell | Adelphia is letting me go quite a bit beyond my rated max |
| 06:02.34 | PTG1234 | hah what are you downloading? :) |
| 06:02.37 | Qwell | I'm getting a stable 600k or so, rated at 5mbit |
| 06:02.41 | Qwell | umm... |
| 06:03.00 | Qwell | erm, rated at 4mbit |
| 06:03.05 | PTG1234 | 5mbit is 625k |
| 06:03.09 | Qwell | yeah |
| 06:03.55 | Qwell | I'm bursting to as much as 675 |
| 06:04.12 | PTG1234 | heh |
| 06:04.15 | PTG1234 | what you getting? |
| 06:04.37 | Qwell | Linux distro! |
| 06:04.40 | PTG1234 | hahaha |
| 06:04.49 | PTG1234 | they have some good porn dvds on there :) |
| 06:04.50 | PTG1234 | i am told |
| 06:04.55 | Qwell | sure, sure |
| 06:05.00 | Qwell | and no, they suck |
| 06:05.01 | Qwell | I hear |
| 06:05.09 | PTG1234 | no way :) |
| 06:05.14 | Qwell | dunno, heh |
| 06:06.16 | DaLion | quiet room |
| 06:07.49 | DaLion | echo 'PTG1234 is thinking about the porn.... Mmmmmm porn' > mail andrea@andreasmail.com |
| 06:07.52 | DaLion | ;) |
| 06:08.01 | Qwell | | |
| 06:08.08 | DaLion | of course the > ouwld be replaced by | |
| 06:08.14 | PTG1234 | haha |
| 06:08.16 | DaLion | Qwell beat me |
| 06:08.17 | DaLion | ;) |
| 06:08.19 | PTG1234 | doubt she would care :) |
| 06:08.26 | Qwell | DaLion: beat you with a with a pipe |
| 06:08.31 | Qwell | hmm |
| 06:08.35 | DaLion | yeah made the mistake once |
| 06:08.40 | DaLion | outpout > file |
| 06:08.43 | DaLion | instead of | |
| 06:08.50 | DaLion | hehe to find my binary overwrot |
| 06:09.04 | DaLion | im playing with the cluster concept of mysql |
| 06:09.09 | DaLion | got 4 boxes to cluster up |
| 06:09.35 | PTG1234 | yah but read only other boxes right? |
| 06:09.42 | DaLion | no |
| 06:09.59 | DaLion | got 1 mgmgt 2 mysql and 2 data nodes |
| 06:10.09 | DaLion | then ill deply mamg and mysql on thery own |
| 06:13.44 | evo4wrx | anyone here had any experience with the crackle on quad span e1 digiums |
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| 06:33.13 | *** join/#asterisk v0ip (v0ip@c-24-19-217-100.hsd1.wa.comcast.net) |
| 06:33.34 | v0ip | hi |
| 06:33.42 | at561 | i just made an industrial strength word speller for phone numbers |
| 06:33.51 | at561 | it includes elite spelling |
| 06:35.04 | v0ip | test |
| 06:36.24 | v0ip | anyone here ever use SS7 for *? |
| 06:37.03 | v0ip | I heard it is possible |
| 06:37.59 | *** join/#asterisk iceyp (~icepick@firewall.unix.co.nz) |
| 06:38.08 | iceyp | has anyone got a calling card platform working with asterisk? |
| 06:38.20 | v0ip | kia ora! |
| 06:38.22 | evo4wrx | voip |
| 06:38.23 | evo4wrx | yes |
| 06:38.23 | iceyp | howdy ho |
| 06:38.26 | iceyp | :) |
| 06:38.43 | iceyp | kio ora! |
| 06:38.44 | *** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com) |
| 06:38.45 | v0ip | evo: how does it work actually? |
| 06:39.02 | v0ip | evo: do you need a leased line? |
| 06:39.07 | *** join/#asterisk dg1nsw (~schulte@gate.sympat.de) |
| 06:39.58 | v0ip | iceyp: wel/akl/chc? |
| 06:40.37 | evo4wrx | u need an interconnect agreement with the other carriers first |
| 06:40.41 | evo4wrx | big bux |
| 06:40.44 | evo4wrx | costs millions |
| 06:41.00 | v0ip | what is the advantage to have SS7 for your * box? |
| 06:41.30 | RoyK | v0ip: direct interconnect, correct billing, blah blah |
| 06:41.40 | RoyK | is there an open ss7 stack available now? |
| 06:42.05 | v0ip | royk: I heard verisign offers SS7 service now |
| 06:42.12 | daork | v0ip: i hope you dont want to do ss7 interconnect in .nz ;) |
| 06:42.57 | v0ip | direct interconnect? what do u mean by that? |
| 06:43.32 | v0ip | daork: who actually "own" ss7 node in nz? :) tnz? clear? vodafone? |
| 06:43.43 | daork | we've got ss7 |
| 06:43.45 | daork | (woosh) |
| 06:43.49 | daork | ihug do |
| 06:43.53 | daork | callplus |
| 06:43.58 | evo4wrx | most major carriers have interconnects |
| 06:44.01 | daork | but its not easy to get |
| 06:44.06 | evo4wrx | its the fastest way to bulk call / bill |
| 06:44.21 | evo4wrx | the company i work for has a carrier lic in Australia and we are looking at getting one |
| 06:44.22 | v0ip | so SS7 is not used only for signalling? |
| 06:44.37 | evo4wrx | to process will cost around 15 milliion to connect to 6 carriers via ss7 |
| 06:45.18 | *** join/#asterisk moua (david@men75-2-82-66-50-159.fbx.proxad.net) |
| 06:45.26 | moua | hello |
| 06:45.36 | evo4wrx | and as far as carrier grade ss7 goes...i wouldnt like to bet any carrier will do it with you on asterisk |
| 06:45.45 | evo4wrx | cause they ask what equipment you use |
| 06:46.00 | evo4wrx | and if you say a open softswitch they will prolly larf |
| 06:46.04 | v0ip | talking about signalling, I still not quite sure how SS7 signalling works ... do they have something like DNS "root servers"? |
| 06:46.21 | evo4wrx | theres an openss7 platform |
| 06:46.24 | evo4wrx | go read that page |
| 06:46.26 | evo4wrx | has heeps on it |
| 06:46.32 | v0ip | url? |
| 06:46.53 | evo4wrx | google it |
| 06:47.10 | v0ip | evo: I heard Cisco has SS7 over IP platform which is cheaper than most SS7 platforms |
| 06:47.22 | cjk | anyone an idea what i need to transcode 8 ilbc to ulaw channels |
| 06:47.51 | moua | before starting with asterisk i'd like to know, can i install asterisk on a server (in console mode) and use a script to call a specific number every 120minutes with a SIP account ? Thanks |
| 06:48.13 | iceyp | daork you in nz? |
| 06:48.34 | daork | iceyp: yes. and you'd be barry whatsisface |
| 06:48.39 | daork | with the isp map thing |
| 06:50.07 | v0ip | daork/iceyp: when does allblack season start? |
| 06:50.27 | iceyp | daryeah |
| 06:50.32 | iceyp | errm daork yes |
| 06:50.47 | iceyp | you working on a pabx for company or for opensource |
| 06:51.09 | daork | v0ip: who knows, i dont care much for rugby |
| 06:51.23 | iceyp | i just watch the Saffas |
| 06:51.37 | v0ip | daork: I thought all kiwis are rugby fans |
| 06:53.39 | *** join/#asterisk elric (~kavit@ppp114-10.static.internode.on.net) |
| 06:54.00 | evo4wrx | we use cisco for ss7 |
| 06:54.49 | iceyp | yeah |
| 06:55.01 | iceyp | woops |
| 06:55.22 | v0ip | evo: itp u mean? |
| 06:55.41 | v0ip | evo: is it stable? cisco kinda new player in this field, don't u think? |
| 06:58.16 | evo4wrx | yea....but we move a lot of cisco product so we have there engineers on call and they are setting it up for free for us |
| 06:58.27 | evo4wrx | cisco are having a lot of problems with HP taking market share right now |
| 06:58.34 | evo4wrx | so they are bending over backwards for people |
| 06:59.24 | v0ip | evo: which product of HP? ProCurves? |
| 06:59.57 | elric | has anyone go any comments regarding aculab cards? |
| 07:00.02 | elric | s/go/got |
| 07:02.32 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 07:03.05 | evo4wrx | not sure of the HP models |
| 07:03.10 | evo4wrx | we havent looked at it yet |
| 07:03.26 | evo4wrx | anyone had anyproblems with crackle on E1's digiums |
| 07:04.11 | yaboo | hp pro services are something to be said |
| 07:04.56 | v0ip | I saw a lot of networkers chose HP ProCurve over Cisco Catalyst simply because of the features available and aggressive pricing |
| 07:08.25 | cypromis | hmmm here cisco mostly goes well cause cisco knows how to help it go through with the right'incetives' in the right 'places' |
| 07:08.40 | cypromis | same goes for HP |
| 07:09.29 | v0ip | cypromis: like free trainings for admins? ;) |
| 07:09.46 | cypromis | nah |
| 07:10.01 | cypromis | since when do Admins sign the cheques to buy anything ? |
| 07:10.16 | *** part/#asterisk RevK_ (~adrianken@81.187.165.154) |
| 07:10.46 | v0ip | sometimes those network admins are the one who influence the purchase ... and the carrots normally are the free trainings :) |
| 07:12.11 | cypromis | not around here |
| 07:12.46 | cypromis | and in most cases the MIS guys have no clue what the difference of a netgear and a extreme switch is |
| 07:12.51 | yaboo | either at work |
| 07:13.12 | v0ip | really? wow |
| 07:14.01 | cypromis | what do you expect ? |
| 07:14.07 | cypromis | proper IT people cost money |
| 07:14.09 | cypromis | hehe |
| 07:14.14 | Romik | any body know how to unlock - ian-02ex the new device from Lingo? |
| 07:14.22 | cypromis | much easier to fire them all and let the right vendor smear your MIS stuff |
| 07:14.43 | v0ip | netgear and extreme is like comparing linksys and juniper :D |
| 07:14.44 | cypromis | they call it outsourcing |
| 07:14.44 | cypromis | lol |
| 07:16.30 | at561 | i can spell 8oil-0a7meal with my phone number |
| 07:16.53 | at561 | awesome |
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| 07:32.15 | int19h | howdy |
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| 07:51.39 | mcnobody | Is anyone using quadbri with bristuff and TE410P/TE405P on same machine? |
| 07:52.16 | *** join/#asterisk W|NGNUT (~wingnut-n@cC30138E9.inet.catch.no) |
| 07:54.18 | daork | at561: if that isnt a claim to fame, i dont know what is |
| 07:55.10 | Romik_ | i do not think this ;possible they for different voltage |
| 07:55.14 | *** join/#asterisk Qwell (~north@24-50-66-194.vnnyca.adelphia.net) |
| 07:55.49 | RoyK | jbot: nickometer at561 |
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| 07:57.43 | daork | jbot: nickometer RoyK |
| 07:57.52 | daork | thats pretty un-lame |
| 07:57.56 | RoyK | :) |
| 07:57.58 | RoyK | mohaha |
| 07:58.23 | daork | heh |
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| 08:03.57 | Qwell | note: Removable IDE drive bays may seem like a good idea...but they will probably die, and kill your drive. |
| 08:10.51 | tengulre | Hi,Qwell! |
| 08:10.55 | *** join/#asterisk zoa (~zoa@pirus.securax.be) |
| 08:11.36 | *** join/#asterisk Martohtar (Martohtar@82.196.218.80) |
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| 08:13.25 | cypromis | Qwell: add the word cheap and it becomes a valid sentence |
| 08:13.48 | Qwell | cypromis: yeah, it was like $12. |
| 08:14.33 | cypromis | you get what you payed for |
| 08:14.41 | Qwell | indeed |
| 08:14.42 | PTG1234 | i doubt it would kill your drive |
| 08:21.59 | *** join/#asterisk Romik (~romik@1.fix.netvision.net.il) |
| 08:22.18 | Micc | What is a good IP Phone to get that works with asterisk for under $100 |
| 08:23.47 | PTG1234 | SIP 841 |
| 08:24.34 | PTG1234 | Sipura |
| 08:26.25 | *** join/#asterisk zione (~zione@81-208-36-80.fastres.net) |
| 08:26.45 | Romik | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=61840&item=5767670702&rd=1&ssPageName=WDVW |
| 08:26.55 | Romik | this one....it supports IAX2 |
| 08:27.15 | PTG1234 | get the sipura 841 |
| 08:27.22 | PTG1234 | if you want to risk having crap get something else :) |
| 08:27.33 | Romik | PTG: did 841 supports IAX2? |
| 08:27.42 | PTG1234 | no and you don't want iax2 |
| 08:27.49 | Romik | ptg: why? |
| 08:27.50 | zoa | you do want iax2 |
| 08:27.50 | PTG1234 | here i'll paste why |
| 08:28.04 | PTG1234 | zoa thats an uneducated statement |
| 08:28.11 | zoa | :) |
| 08:28.15 | Qwell | PTG1234: Before I took the fan out a year or so ago, it wasn't pushing enough power to it, causing it to die...loudly. |
| 08:28.16 | PTG1234 | <PROTECTED> |
| 08:28.16 | PTG1234 | This assumes everything is same except protocols used. |
| 08:28.16 | PTG1234 | <PROTECTED> |
| 08:28.16 | PTG1234 | Normal telephone call with iax |
| 08:28.16 | PTG1234 | LEVEL3(CA) -> VOIP_WHOLESALE_PROVIDER_SWITCH(CO) -> VOIP_WHOLESALE_PROVIDER_SERVER(CO) -> YOUR_PROVIDER(TX) -> YOUR_SERVER(NY) -> YOUR_CUSTOMER(WA) |
| 08:28.18 | PTG1234 | <PROTECTED> |
| 08:28.20 | PTG1234 | Normal telephone call using same providers with SIP |
| 08:28.22 | PTG1234 | LEVEL3(CA) -> YOUR_CUSTOMER(WA) |
| 08:28.24 | PTG1234 | <PROTECTED> |
| 08:28.26 | PTG1234 | call #1 150ms |
| 08:28.26 | Qwell | PTG1234: then it was fine for a while, and it started being loud and stupid yet again |
| 08:28.28 | PTG1234 | call #2 20ms |
| 08:28.30 | PTG1234 | <PROTECTED> |
| 08:28.32 | PTG1234 | - in SIP they will all negociate so RTP streams will go direct |
| 08:28.34 | PTG1234 | - in IAX they can negociate somewhat but since both ends are SIP its sort of pointless |
| 08:28.36 | PTG1234 | - until all phones use IAX and L3 uses IAX (Which won't happen) SIP is the better choice |
| 08:28.38 | PTG1234 | <PROTECTED> |
| 08:28.42 | PTG1234 | ALso... |
| 08:28.44 | PTG1234 | <PROTECTED> |
| 08:28.46 | PTG1234 | <PROTECTED> |
| 08:28.48 | PTG1234 | Most VOIP providers receive everything in SIP, so why would you want an added point of failure, that asterisk box in the middle? |
| 08:28.53 | PTG1234 | Qwell: hmm weird maybe it could |
| 08:28.55 | PTG1234 | Qwell: i guess i could be wrong |
| 08:28.57 | PTG1234 | romik: understand? |
| 08:30.05 | Romik | ptg: reading |
| 08:30.49 | Micc | Ok so where can I get a Sipura 841 for a good price? |
| 08:30.56 | PTG1234 | voipsupply.com |
| 08:31.09 | PTG1234 | as a rule of thumb just set your phone to reinvite every 2 minutes |
| 08:31.15 | PTG1234 | and you will have no firewall problems |
| 08:31.17 | Romik | PTG: what the best way to connect to wholesale provider - SIP or IAX of they support both? |
| 08:31.19 | Qwell | off to bed...not sure why I'm still up |
| 08:31.24 | PTG1234 | romik: sip |
| 08:31.27 | PTG1234 | always use sip :) |
| 08:31.30 | PTG1234 | qwell: good night |
| 08:31.35 | dec | really? wow. |
| 08:31.43 | dec | even when connecting from an asterisk box to the provider? |
| 08:31.48 | PTG1234 | <PROTECTED> |
| 08:31.48 | PTG1234 | <PROTECTED> |
| 08:31.58 | PTG1234 | why do this |
| 08:32.03 | RoyK | http://karlsbakk.net/advice/ |
| 08:32.07 | PTG1234 | SIP->ASTERISK->ASTERISK |
| 08:32.10 | PTG1234 | SIP->ASTERISK |
| 08:32.12 | PTG1234 | makes more sense |
| 08:32.21 | dec | fair enough |
| 08:32.24 | dec | thanks |
| 08:32.26 | PTG1234 | really you should use SER + ASTERISK for now |
| 08:32.42 | dec | yeah i'm going to setup SER soon |
| 08:32.53 | PTG1234 | eventually i will release a new chan_Sip that will proxy like ser |
| 08:32.56 | PTG1234 | then ser won't be needed |
| 08:33.42 | dec | that'd be great |
| 08:33.43 | dec | can't wait |
| 08:33.58 | Romik | i have following setup office with 120 phones with channel banks + Digium quad cards (iax2, israel) --speex-170ms-->PBX (new york, IAX2)-1ms-ulaw--> Whole sales provider (New york) |
| 08:34.26 | PTG1234 | first of all i wouldn't use speex |
| 08:34.53 | PTG1234 | who's your provider? |
| 08:34.54 | RoyK | speex is a really nice way of wasting cpu..... |
| 08:35.00 | Romik | livevoip + voipjet |
| 08:35.12 | PTG1234 | well its not that, its that its not widely supported.. and if you don't tweak it right you can have problems |
| 08:35.26 | RoyK | anyone tried VoIP over ISDN dialup? |
| 08:35.28 | RoyK | :P |
| 08:35.31 | PTG1234 | livevoip you would want to hand off with SIP.. if you do they put you right on like TNTs and good equipment.. if you don't you get put on a shitty pc |
| 08:35.40 | PTG1234 | if you run sip through the whole route |
| 08:35.43 | PTG1234 | enable reinvite |
| 08:35.47 | PTG1234 | your quality will go up |
| 08:35.52 | PTG1234 | also use g729 the whole way |
| 08:36.01 | PTG1234 | then you should have calls go direct to voipsupply |
| 08:36.06 | PTG1234 | they will only initially use your box |
| 08:36.10 | PTG1234 | but then pull it out of the loop |
| 08:36.11 | Romik | PTG: i tested g729 and speex looks like speex much better quality |
| 08:36.37 | PTG1234 | romik: its not.. |
| 08:36.46 | PTG1234 | romik: not to mention you need to use the same codec all the way through |
| 08:36.49 | Romik | PTG: like israel->newyork is very jitterered |
| 08:36.50 | PTG1234 | so all devices have to support it |
| 08:36.59 | PTG1234 | so on the most part that limits you to G729a and ULAW |
| 08:37.12 | PTG1234 | Romik: it shouldn't be, but you could use the new jitterbuffer |
| 08:37.29 | PTG1234 | thats most likely livevoips and voipjets crappy routes |
| 08:37.39 | Romik | PTG: both my PBX support speex, ulaw, g729 |
| 08:37.40 | PTG1234 | i would never consider using either of them for business class service |
| 08:37.43 | *** join/#asterisk clive- (~pirch@rndf-146-44-91.telkomadsl.co.za) |
| 08:37.48 | PTG1234 | romik: but your PHONES don't |
| 08:37.52 | PTG1234 | they have to go all the way through |
| 08:38.04 | PTG1234 | or are you using analog phones? |
| 08:38.05 | Romik | PTG: tell me somebody better |
| 08:38.08 | Romik | PTG: yes |
| 08:38.35 | PTG1234 | well g729 uses less bandwidth.. so it should be better for jitter issues |
| 08:39.15 | Romik | PTG: but current g729 is not support new jitterbuffer |
| 08:39.29 | PTG1234 | are you sure, i thought steve says it is |
| 08:39.36 | Romik | PTG: as i undestand from this channel...Mark need to recompile it |
| 08:39.44 | PTG1234 | ask stevek |
| 08:39.57 | PTG1234 | but i just can't imagine jitter issues |
| 08:40.02 | PTG1234 | whats the ip of the box in israel? |
| 08:40.14 | Romik | ptg: what do you mean? |
| 08:40.26 | Romik | ptg: "ip of the box" ? |
| 08:40.30 | *** part/#asterisk sympad (~Misha@195.138.127.98) |
| 08:40.38 | PTG1234 | yah um.. |
| 08:40.41 | Micc | PTG, is nufone.net a pretty good sip provider? |
| 08:40.51 | PTG1234 | <PROTECTED> |
| 08:40.53 | PTG1234 | that link |
| 08:40.54 | PTG1234 | whats the ip |
| 08:40.57 | PTG1234 | i want to see your route |
| 08:41.03 | Romik | ah... |
| 08:41.22 | clive- | micc nufone uses asterisk, which does not do jitter buffering on sip at all...(yet), but if you are on a good connection, they are fine |
| 08:41.22 | Romik | ptg: i am firewalled....it useless.. give me your ip i will traceroute |
| 08:41.32 | PTG1234 | sip1.way2fast.com |
| 08:41.54 | Micc | What is the best SIP provider? |
| 08:42.20 | PTG1234 | i wouldn't use nufone, but thats me.. and i wouldn't think anyone ever is gonna badmouth them really in this channel |
| 08:42.21 | tainted- | hey PTG1234 any luck? |
| 08:42.41 | PTG1234 | hey tainted :) sorry been swapped lately.. talk to me tommorow we can discuss the details |
| 08:42.51 | PTG1234 | been implementing a new dial module |
| 08:43.01 | Micc | PTG, so who would you use? |
| 08:43.04 | PTG1234 | and backporting everything i have to stable |
| 08:43.17 | PTG1234 | it depends on usage, if its a business class service, etc |
| 08:43.28 | PTG1234 | i wouldn't use anyone who is gonna connect me to a server |
| 08:43.32 | Romik | ptg: http://pastebin.ca/9979 |
| 08:43.36 | PTG1234 | your likely to have problems |
| 08:43.59 | PTG1234 | 200ms to the first hop in israel? |
| 08:44.13 | PTG1234 | you got some serious internal issues |
| 08:45.12 | PTG1234 | then you add another 200ms in israel |
| 08:45.16 | Romik | ptg: there was problem : look at end http://pastebin.ca/9980 |
| 08:45.28 | Micc | PTG, who's got the best service. They all seem pretty cheap to me. Broadvoice seems to have some issues sometimes. |
| 08:45.49 | PTG1234 | romik: are they both from the same box? |
| 08:46.02 | Romik | ptg: there was some temporary problem yes... |
| 08:46.17 | *** join/#asterisk saabluvr (master@keeper.nc-ks.de) |
| 08:46.17 | Romik | ptg: your server routed via london |
| 08:46.19 | PTG1234 | see your problem def. is your connection isn't consistent |
| 08:46.39 | PTG1234 | do this |
| 08:46.42 | PTG1234 | ping me with 50 packets |
| 08:46.44 | PTG1234 | and paste |
| 08:47.06 | PTG1234 | btw 150ms isn't bad considering your going to west coast |
| 08:47.08 | Romik | 243ms |
| 08:47.10 | PTG1234 | but its not consistent |
| 08:47.18 | PTG1234 | i wanna see all 50 results :) |
| 08:47.27 | Romik | 242-263ms |
| 08:47.29 | Romik | w8 |
| 08:48.00 | *** part/#asterisk dg1nsw (~schulte@gate.sympat.de) |
| 08:48.04 | Romik | http://pastebin.ca/9981 |
| 08:48.14 | PTG1234 | Micc: what type of usage? |
| 08:48.19 | PTG1234 | micc: just general home use or what? |
| 08:48.40 | PTG1234 | wow that is horrible |
| 08:48.53 | clive- | when is version 1.2 expected? |
| 08:48.57 | PTG1234 | now paste me a ping to your newyork network |
| 08:49.05 | PTG1234 | 50 packets |
| 08:49.19 | PTG1234 | btw speex has its own jitterbuffer type stuff, which is why it may work better for you |
| 08:49.29 | Romik | http://pastebin.ca/9982 |
| 08:49.43 | PTG1234 | wow |
| 08:49.46 | PTG1234 | that is horrible |
| 08:49.54 | PTG1234 | you basically need a jitter buffer st at 600ms |
| 08:50.09 | PTG1234 | thats worse then a sattelite link |
| 08:50.24 | Micc | PTG, both. I like asterisk so much I'll be setting it up for myself and friends, but I've got to set it up for my work too. |
| 08:50.27 | Romik | to new york it's OK...160ms... |
| 08:50.35 | PTG1234 | no its not |
| 08:50.41 | PTG1234 | 034 64 bytes from 66.246.222.72: icmp_seq=31 ttl=57 time=421.4 ms |
| 08:50.41 | PTG1234 | 035 64 bytes from 66.246.222.72: icmp_seq=32 ttl=57 time=399.3 ms |
| 08:50.42 | PTG1234 | 036 64 bytes from 66.246.222.72: icmp_seq=33 ttl=57 time=578.6 ms |
| 08:50.42 | PTG1234 | 037 64 bytes from 66.246.222.72: icmp_seq=34 ttl=57 time=574.9 ms |
| 08:50.42 | PTG1234 | 038 64 bytes from 66.246.222.72: icmp_seq=35 ttl=57 time=414.4 ms |
| 08:50.42 | PTG1234 | 039 64 bytes from 66.246.222.72: icmp_seq=36 ttl=57 time=156.4 ms |
| 08:50.45 | PTG1234 | thats what causes your issues |
| 08:50.55 | PTG1234 | its consistency.. voip doesn't need a fast connection, but a consistent one |
| 08:51.01 | clive- | south africa on a good day is 400ms to NY, used to be like 700 before when they used satelite, and voip still worked |
| 08:51.23 | PTG1234 | if your packets always take 700ms, its gonna work well :) |
| 08:51.26 | Romik | PTG: i will speak with ISP to get some qos |
| 08:51.27 | PTG1234 | you may be a second lagged |
| 08:51.29 | PTG1234 | but it will work well |
| 08:51.40 | PTG1234 | romik: you need to identify what is causing the inconsistency |
| 08:51.43 | clive- | yes, the delay is noticable |
| 08:51.46 | PTG1234 | my guess, overworked router |
| 08:51.50 | Micc | PTG, for work we're looking at at least 20 lines inbound with most of those being in use at any given time. |
| 08:52.03 | PTG1234 | micc: what area codes? |
| 08:52.27 | Romik | ptg: any advice for me?:) |
| 08:52.44 | PTG1234 | romik: ping each point in that traceroute with 50 packets |
| 08:52.52 | Micc | PTG, we'll probably be using some 800 numbers and the others don't matter. |
| 08:52.52 | PTG1234 | find which is the first one that cuases the inconsistency |
| 08:53.04 | PTG1234 | micc: ok so all 800, and outbound.. no local needed? |
| 08:53.10 | Micc | PTG, we'll do a hunt group from an 800 number. |
| 08:53.24 | PTG1234 | any idea on monthly usage? |
| 08:53.38 | Micc | PTG, Our users are all over the US and very few in other countries. |
| 08:54.00 | PTG1234 | micc: how many minutes per month? |
| 08:54.05 | PTG1234 | and this is mostly for a business then? |
| 08:54.06 | Romik | currently : we have 500min a day but half comes from Israel and half from Antigua |
| 08:54.29 | Romik | ptg: we have a problems...we can get easy 1000min/day when it will work OK |
| 08:55.07 | PTG1234 | heh i was talking to micc :).. you need to fix your connection romik more then anything |
| 08:55.16 | PTG1234 | can you do the 50packet thing |
| 08:55.18 | PTG1234 | paste me each one |
| 08:55.20 | PTG1234 | one by one |
| 08:55.27 | PTG1234 | start with the first in the link |
| 08:55.58 | PTG1234 | romik: where are most of your customer located? |
| 08:56.06 | Micc | PTG, yes business. I figure at least 10% usage of the 20 lines. thats 43200 minutes. |
| 08:56.28 | *** join/#asterisk _andi (~andi@numenor.segfault.net) |
| 08:56.42 | Romik | ptg: US mostly but also spain, france, italia, germany - we route our french traffic via acropolis (french voip) |
| 08:56.44 | Micc | That would be just for our first rollout. If we start doing conferencing we could use 400 lines at any given time. |
| 08:56.51 | Romik | ptg: japan also |
| 08:56.56 | _andi | i've problems dialingout with digium te110p ( Ext: 1 Cause: Info. element nonexist or not implemented (99), class = Protocol Error (6) ]) |
| 08:56.58 | PTG1234 | micc: pm me real quick |
| 08:57.10 | _andi | kernel modul asterisk etc is running fine because I can accept calls coming in |
| 08:57.11 | PTG1234 | romik: and your using livevoip? |
| 08:58.03 | Romik | ptg: yes. the best i found out. |
| 08:58.21 | *** join/#asterisk Xander77 (~Alex@exten-halls-243.soton.ac.uk) |
| 08:59.19 | PTG1234 | you had many problems with them like alot of others have told me? |
| 08:59.26 | Romik | ptg: most calls go via livevoip, but like antigua cells is better to route via voipjet |
| 08:59.49 | PTG1234 | yah i don't think any of those guys concentrate on getting better rates in better areas |
| 08:59.49 | Romik | ptg: huge qantity of problems...becouse i am here...;) |
| 08:59.55 | PTG1234 | they just all route everything to one provider |
| 09:00.06 | PTG1234 | a good provider you shouldn't need ot be able to switch around |
| 09:00.29 | PTG1234 | romik: do what i say with the pings, i am going to bed soon if you want me to fix your problems |
| 09:00.32 | Romik | ptg: tell me about one. |
| 09:00.50 | saabluvr | Hi ! Can ANYONE confirm that spandsp is supposed to work with zaphfc ? |
| 09:00.56 | Romik | ptg: what do you mean? |
| 09:01.09 | PTG1234 | romik: private message me so i don't have to help you in the open channel :) it gets confusing |
| 09:01.11 | Romik | ptg: i will speak with ISP today..regarding consistancy |
| 09:01.19 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 09:04.57 | saabluvr | is noone running rxfax with zaphfc ? |
| 09:07.39 | saabluvr | wow. Didn't think this combination would be so exotic |
| 09:08.00 | saabluvr | Is there another solution to receive faxes with zaphfc ? |
| 09:14.40 | *** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au) |
| 09:15.37 | *** join/#asterisk FaithX (~Faithful@202-6-145-116.ip.adam.com.au) |
| 09:18.52 | tainted- | looks like voicepulse changed their callerid format |
| 09:21.36 | *** join/#asterisk three55ml (~three55ml@cpe-66-68-98-68.austin.res.rr.com) |
| 09:21.45 | three55ml | Hello all |
| 09:22.19 | PTG1234 | hey three |
| 09:22.29 | PTG1234 | ok i have played oracle enough for tonight :) its time to go to bed |
| 09:23.00 | three55ml | I've played poker enough tonight, it's time to do some work :) |
| 09:23.39 | PTG1234 | i only like poker in person.. you can;'t tell if the person is bluffing through a pc :) |
| 09:24.17 | three55ml | I played a live game tonight, I play a lot online though as well. |
| 09:32.24 | *** join/#asterisk Chickens (~NNSCRIPT@70.56.190.138) |
| 09:33.35 | *** join/#asterisk Manipura (~chatzilla@dsl-ep-209-115-250-i114-cgy.nucleus.com) |
| 09:33.59 | Manipura | is there anyway to log in a file or a database what goes on in the cli? |
| 09:39.47 | Makenshi | is it possible to record sip calls when using asterisk as an rtp proxy? |
| 09:43.21 | *** join/#asterisk pino (~z@host39-28.pool21345.interbusiness.it) |
| 09:44.41 | tessier | Makenshi: Yes |
| 09:51.47 | *** join/#asterisk ptg123 (~ptg123@h460601b4.area1.spcsdns.net) |
| 09:52.08 | ptg123 | heh on pda irc from bed |
| 09:53.05 | Manipura | Thats a junkie |
| 09:53.34 | ptg123 | i had to try it |
| 09:54.43 | ptg123 | man this thing is the best spent money ever |
| 09:55.16 | ptg123 | damn bill is a smart guy |
| 09:58.08 | *** join/#asterisk Manipura (~chatzilla@dsl-ep-209-115-250-i114-cgy.nucleus.com) |
| 09:58.55 | syle | kewl glad your happy |
| 09:59.18 | syle | mine was my eye laser surgery, best investment i ever made, something i use everyday as well :) |
| 10:01.24 | ptg123 | i have 20/10 vision, so no need for that |
| 10:05.27 | three55ml | ptg123: What do PDA do you have? |
| 10:05.50 | three55ml | I have an e740 but it sucks now because Toshiba didn't offer the 2003 update. It was cool a few years ago though. Now it's a paperweight. |
| 10:06.52 | syle | can i talk seriously for a sec |
| 10:07.41 | syle | i am basically in a situation now where i need to start a new company.... |
| 10:07.53 | syle | i was thinking seriously about selling voip |
| 10:08.00 | syle | is it profitable or not? |
| 10:08.48 | three55ml | What do you want to sell...and to whom? |
| 10:08.49 | syle | i mean if you charge $19.99 unlimited , how much of that goes in your pocket from like sip peering with level3.com or whatever |
| 10:09.12 | three55ml | You need some change up front to get the agreements with Level 3, Broadvox, etc. if you want to jump straight into that. |
| 10:09.18 | tengulre | I have a card (support SS7) but the asterisk can't support it, who can help me see it's head file? |
| 10:09.48 | tengulre | SOS! |
| 10:10.02 | syle | i got some change up front, that is not worries me, its more if it will work or not |
| 10:10.25 | cypromis | depends how good you are in sales and keeping your costs down |
| 10:10.31 | cypromis | as in every other business as well |
| 10:10.31 | three55ml | syle: It's all a numbers game with unlmited accounts. You're banking on the fact most of your users won't ever use that many minutes to put you in the red for their account, but a few will...so you have to make sure you have enough who use less than your cost so in the end it all averages out. |
| 10:10.35 | syle | i can throw down $50k tommorrow on a business so not that much but a good amount |
| 10:10.47 | three55ml | tengulre: Sorry, no experience with it |
| 10:10.54 | Manipura | ~seen twk-b |
| 10:10.57 | jbot | Manipura: i haven't seen 'twk-b' |
| 10:11.16 | tengulre | give me your email! |
| 10:11.25 | tengulre | I send it to you |
| 10:11.43 | syle | i see, i was thinking starting with companies would be a good start, then offer residential as well |
| 10:11.53 | three55ml | syle: More than likely, if you go striaght to the big guys...you'd piss away most of that in a few months just on minimum commitments (at least from numbers I've gotten.) |
| 10:12.38 | three55ml | Consumer is a hard market to crack into with all the cable companies and phone companies rolling it out. Hard to compete with that kind of existing customer base to market to, not to mention billion dollar marketing budgets. |
| 10:12.55 | three55ml | At least in the US |
| 10:13.27 | syle | very true i suppose, it is very competitive , sure AOL etc will capture that market, but if you just got 5% of that pie would it be worthwhile? |
| 10:13.30 | *** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de) |
| 10:13.40 | Chickens | most of the large cable companies are putting in VoIP systems for home and business users as we speak. |
| 10:13.44 | *** join/#asterisk moua (david@men75-2-82-66-50-159.fbx.proxad.net) |
| 10:13.52 | moua | before starting with asterisk i'd like to know, can i install asterisk on a server (in console mode) and use a script to call a specific number every 120minutes with a SIP account ? Thanks |
| 10:14.04 | tengulre | three55ml,can you help me ? I have the hardware's documents and the hardware support linux! |
| 10:14.14 | three55ml | tengulre: No, sorry |
| 10:14.23 | cypromis | syle: you expect 5% of the market with a budget of 50k ? |
| 10:14.29 | tengulre | tree55ml,pls! |
| 10:14.34 | three55ml | tengulre: Try #asterisk-dev, but maybe tommorow morning (US time) would be a better time to reach people |
| 10:14.37 | syle | so maybe i am better off just opening a bar locally here lol, problem is i hate people and like computers lol |
| 10:14.40 | cypromis | that's not even enough for an initial marketing campaign |
| 10:15.00 | tengulre | three55ml,tks |
| 10:15.04 | three55ml | syle: 50k won't open a bar either :) At least not here. Bar's have very small profit margins in most instances anyways. |
| 10:15.10 | Chickens | start a consulting company or something |
| 10:15.10 | cypromis | to run that kind of business you should love sales |
| 10:15.13 | cypromis | and not computers |
| 10:15.14 | syle | my marketing campaign would consist of some very low cost telemarkets from canada |
| 10:15.18 | moua | wich market ? Liechtenstein ? :) |
| 10:15.27 | cypromis | rather san marino |
| 10:16.14 | cypromis | syle: try to sell your know how to one of the guys that want to do a mid size roll out |
| 10:16.18 | syle | well money is about marketing period obviously everyone knows that, i guess i am just looking for advice based on my situation |
| 10:16.50 | cypromis | as in you could outsource their voip |
| 10:17.08 | cypromis | than you do basically the same as you wanted but somebody else covers the risk |
| 10:17.09 | cypromis | hehe |
| 10:17.15 | *** join/#asterisk W|NGNUT (~wingnut-n@128.80-203-103.nextgentel.com) |
| 10:17.18 | Chickens | how would you do QoS and all that kind of stuff for them? |
| 10:17.44 | cypromis | the same way aas an inhouse department |
| 10:17.54 | cypromis | nobody said you should offer them a hardware outsource |
| 10:17.58 | cypromis | just a now how outsource |
| 10:18.10 | syle | i want to avoid that though cypromis that always ends up in a stable job environment situation, and i want it to be my own company this time :) |
| 10:18.59 | Chickens | create a consultant firm or something like that |
| 10:19.19 | newl | You'll need heaps more for startup and operating capital then. :) |
| 10:19.45 | newl | $50k would pay your salary for one year. What about the others? |
| 10:20.00 | syle | how much you need for capital, i figured one good unix server running asterisk->sip peering to another voip provider and reselling that way |
| 10:20.20 | cypromis | a voip business is like, throw money out of the window for 1-3 years |
| 10:20.29 | cypromis | and pray you will make profit afterwards on your customer base |
| 10:20.30 | cypromis | lol |
| 10:20.31 | syle | seriously |
| 10:20.36 | syle | ouch |
| 10:20.42 | three55ml | Not necessarily |
| 10:21.05 | cypromis | depends of course how you rate your hours |
| 10:21.17 | three55ml | Well it depends on hundreds of variables |
| 10:21.26 | newl | As does profitability. |
| 10:21.47 | three55ml | syle: Are you in the US? |
| 10:21.54 | syle | canada |
| 10:22.08 | Manipura | I just tried updating * and it messed up on asterisk-addons. So now I tried re-installing the previous cvs I had running and I'ts still not working |
| 10:22.15 | syle | for last 3 years i have been working out of US then AOL put a stop to my business unfortunately |
| 10:22.41 | syle | why i am here looking for new ways to make money i suppose |
| 10:22.41 | three55ml | As in it was a division of AOL or you were spamming on AOL? :) |
| 10:23.03 | Manipura | Now when I try to start *, it can't load all these modules. I turn the mod. off in the conf and it just keep's going saying it can't load the next module. |
| 10:23.12 | syle | not technically spam if you follow can-spam acts, but they slap you with lawsuits regardless, so cannot continue |
| 10:23.14 | newl | the pay was nice, the hours sucked, then the IT market really dried up, especially with all the kiddies coming out of school thinking they know it all. :) |
| 10:23.15 | cypromis | one thing I think could generate some profit |
| 10:23.23 | cypromis | but involves a lot of direct customer contact |
| 10:23.30 | cypromis | are the small area services |
| 10:23.35 | three55ml | I own several companies, one of which nets several million a year...and being the boss still isn't that fun 99% of the time :) |
| 10:23.45 | cypromis | as in you provide services for customers close to your lcoalisation |
| 10:24.09 | syle | several million a year nice, all legit? |
| 10:24.18 | Manipura | Anyone know if these modules are needed or what they are needed for? app_md5.so,app_readfile.so,app_chanspy.so,cdr_custom.so |
| 10:24.26 | syle | i was working on my first million then these lawsuits happened :( |
| 10:24.31 | syle | was i ever pissed |
| 10:24.31 | newl | three55ml: That's the time to hire someone to play boss for you. hehe |
| 10:24.41 | three55ml | syle: Yes, the one in parituclar that does very well I only own ~25% of though. |
| 10:25.08 | syle | very nice |
| 10:25.19 | evo4wrx | you would have to be nuts to want to own a VOIP company |
| 10:25.24 | evo4wrx | you have serious thrill issues |
| 10:25.33 | cypromis | hahahahaha |
| 10:25.53 | cypromis | yeah that is what I tell customers who come to us to become voip companies as well |
| 10:25.56 | cypromis | but they insist |
| 10:25.59 | cypromis | lol |
| 10:26.04 | evo4wrx | yea |
| 10:26.06 | evo4wrx | no shit |
| 10:26.12 | evo4wrx | ISP's love it |
| 10:26.21 | cypromis | everybody loves it |
| 10:26.25 | *** join/#asterisk _THEEND_ (~DrEaM@host37-42.pool8248.interbusiness.it) |
| 10:26.27 | cypromis | last one that came was a pharmacist |
| 10:26.28 | evo4wrx | yea |
| 10:26.35 | evo4wrx | its not hard to sell it as a solution |
| 10:26.39 | newl | It would've been good to get into a year ago but now that alot of the ILECs, CLECs and the like are gearing up, the writing is on the wall. The big will survive. :) |
| 10:26.43 | _THEEND_ | who has nat rtp ports on cisco router? |
| 10:26.58 | cypromis | newl: 5 years ago |
| 10:27.05 | evo4wrx | the company i work for is a massive voip company |
| 10:27.09 | cypromis | now is really late |
| 10:27.09 | evo4wrx | so we are here to stay |
| 10:27.12 | evo4wrx | :) |
| 10:27.22 | newl | cypromis: I live on the ass of the planet. Things here are 5-10 years behind. hehe |
| 10:27.31 | cypromis | where is that ? |
| 10:27.34 | three55ml | newl: Kansas? |
| 10:27.35 | cypromis | I am in central europe |
| 10:27.38 | cypromis | :P |
| 10:27.40 | newl | Australia |
| 10:27.43 | cypromis | hehe |
| 10:27.46 | evo4wrx | oi |
| 10:27.49 | evo4wrx | i wouldnt be saying that |
| 10:27.49 | cypromis | the place where everything costs double ? |
| 10:27.50 | cypromis | lol |
| 10:27.58 | evo4wrx | im from Aust |
| 10:28.07 | *** join/#asterisk pigpen (~mark@fw.seamans.cc) |
| 10:28.08 | newl | evo4wrx: hah WA too |
| 10:28.09 | jonathh | hello one and all.. quick question.. any special kernel options needed to get the xp100 clone card working? |
| 10:28.13 | evo4wrx | you got it |
| 10:28.33 | newl | evo4wrx: being in WA, you'd know it to generally be true too. |
| 10:28.57 | pigpen | jonathh: to my knowledge only if you are running module-less... |
| 10:29.10 | evo4wrx | yea |
| 10:29.22 | evo4wrx | but the company i work for is in Sydney and melbourne too |
| 10:29.22 | jonathh | i am about strip my kernel down to what i see as bare minimals.. |
| 10:29.29 | evo4wrx | so the wa side is slower |
| 10:29.30 | evo4wrx | :P |
| 10:29.53 | pigpen | jonathh: cool...I have been doing that for about 3 years....using gentoo. |
| 10:30.21 | jonathh | good man.. we use gentoo at work.. but i am using feddora at home.. so i get a good mix of skills. |
| 10:30.22 | _THEEND_ | who has nat rtp ports on cisco router? |
| 10:30.40 | jonathh | this is my first kernel compile.. kinda exciting :) |
| 10:30.53 | pigpen | cool...enjoy... |
| 10:31.00 | three55ml | A shameless plug here, but I'm going to release this in the next few days - http://www.premierpbx.com/index_new.php - any comments? |
| 10:31.01 | jonathh | hehe.. i'll be back i am sure |
| 10:31.06 | pigpen | we are gentoo devs...so we compile alot. |
| 10:31.10 | three55ml | jonathh: Have fun with that one :) |
| 10:31.38 | newl | evo4wrx: There's a big voip company over east? Which one? Don't say Telstra becuase I know better. :) |
| 10:32.00 | jonathh | yeah gentoo is dead brill.. but i dont know enough of the stuff in the middle.. so i am opting for a distro other than that at home.. so i get more hands on |
| 10:32.03 | evo4wrx | well |
| 10:32.05 | evo4wrx | theres engin |
| 10:32.12 | evo4wrx | theres techex |
| 10:32.16 | evo4wrx | and ipsystems |
| 10:32.21 | evo4wrx | only ones worth talking about |
| 10:32.31 | newl | Engin was the first one that popped into my head. I've never heard of the other two. |
| 10:33.17 | evo4wrx | yea |
| 10:33.24 | evo4wrx | not many peopel knwo about the other ones |
| 10:33.30 | evo4wrx | there is comindico...they went bust |
| 10:33.32 | evo4wrx | they are crap |
| 10:33.36 | newl | Won't be too much longer before the borg make their voip products available to their half a million customers. |
| 10:33.45 | evo4wrx | then there is a bigger one again...not then comindico there like an engin |
| 10:33.47 | newl | heh comindico yeah, agree with ya on that. |
| 10:33.51 | evo4wrx | but they dont even tell anyone there around |
| 10:33.54 | pigpen | any word how nufone is doing with their upgrades? |
| 10:33.58 | evo4wrx | they have massive big corporates ont here network |
| 10:34.03 | *** join/#asterisk RestLessGemini (~umairbari@202.142.189.86) |
| 10:34.25 | evo4wrx | but you arnt allowed on there network unless you sign an NDA not to tell anyone about them |
| 10:34.26 | newl | Involved with the SA state government by any chance? 8) |
| 10:34.29 | evo4wrx | they are hiding from telstra |
| 10:34.45 | evo4wrx | they are good mates with a lot of the senators |
| 10:34.47 | clive- | newl I am in SA, but not in the governemnt |
| 10:34.48 | clive- | lol |
| 10:35.22 | evo4wrx | they will be the first VOIP to get SS& in aust too |
| 10:35.34 | evo4wrx | word has it they are about to roll out a GSM/Wifi seamless voip network too |
| 10:36.04 | *** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com) |
| 10:36.27 | clive- | evo4wrx where is that? |
| 10:36.29 | newl | We recently had this rahrah meeting recently about voip and how BT were doing it already. We were even given pretty black Cisco phones that we're meant to support but alas nobody trained us (properly, but they're not exactly hard). |
| 10:36.43 | cypromis | clive-: australia |
| 10:36.54 | clive- | thanks MIke |
| 10:37.02 | cypromis | np |
| 10:37.14 | clive- | does anyone know when 1.2 version of * will be released |
| 10:37.34 | cypromis | any other jokes of the day ? |
| 10:37.38 | cypromis | :P |
| 10:37.50 | clive- | did I say something funny?..:) |
| 10:37.58 | evo4wrx | well that company is also doing a deal with netxusa |
| 10:38.00 | evo4wrx | and vonage |
| 10:38.03 | evo4wrx | and maybe chitel |
| 10:38.16 | evo4wrx | now thats a thrill issue |
| 10:38.28 | newl | heh |
| 10:39.10 | _andi | anybody using digium te110p ? |
| 10:39.43 | newl | I really miss quite a few creture comfort things from home, such as DTV, my TiVO, cheap hardware (of all types), and cars that don't cost $40k+ AUD. |
| 10:42.53 | kajtzu | _andi: yes |
| 10:44.15 | tengulre | does the digium te110p support chinese PSTN? |
| 10:50.45 | *** join/#asterisk zotz (~zotz@24.231.32.109) |
| 10:51.16 | *** join/#asterisk znoG (gs@200.115.216.109) |
| 10:51.35 | *** join/#asterisk Nuttah (~andrew@amber.interdart.co.uk) |
| 10:54.40 | TUplink | i have some old dialogic cards here will they work with asterisk? |
| 10:54.52 | cypromis | If you buy a driver from digium |
| 10:54.53 | cypromis | they will |
| 10:55.12 | TUplink | have to buy the driver? |
| 10:55.17 | cypromis | dunno |
| 10:55.29 | cypromis | I suppose contacting sales@digium.com could help |
| 10:55.41 | TUplink | someone bought the hardware driver should be included in that |
| 10:56.03 | cypromis | ? |
| 10:56.13 | cypromis | it's not relatex |
| 10:56.16 | cypromis | related even |
| 10:56.33 | TUplink | its like buying a car and getting no fenders |
| 10:56.41 | tainted- | no it's not |
| 10:56.41 | cypromis | don't think intel is interested in writing drivers for every piece of softwrae around |
| 10:56.59 | cypromis | you don't get a petrol station for free when you buy a car |
| 10:57.06 | tainted- | it's like buying a car and expecting the road to be paved wherever u want to go |
| 10:57.33 | cypromis | yeah or buying a truck and expecting MAC to provide for the goods to be transported as well |
| 10:58.14 | tainted- | or like like buying a card and expecting drivers for every platform / software application |
| 10:58.36 | cypromis | hehe |
| 10:59.02 | TUplink | i dont buy it if it willnot work onhte platform that i want it to |
| 10:59.09 | TUplink | oh hehe listen to this |
| 10:59.10 | tainted- | if that were the case, half of the reverse engineered drivers wouldn't need to exist |
| 10:59.19 | tainted- | u said old cards |
| 10:59.39 | cypromis | Intel will cry |
| 10:59.46 | cypromis | if you don't buy their cards |
| 10:59.47 | cypromis | lol |
| 11:00.36 | tengulre | :) |
| 11:00.41 | TUplink | in my begining days of *nix i bought a NIC from circuit city was a netgear FA310TX on hte box it says linux and windows... so i get it.... get it home and cant get it t owork... so i call netgear and they tell me they dont support redhat... |
| 11:00.48 | TUplink | i did figure it out tho |
| 11:01.03 | tainted- | there are linux hardware compatibility lists |
| 11:01.17 | TUplink | that card dose work |
| 11:01.29 | TUplink | its a dc0 card |
| 11:01.40 | tainted- | mkay |
| 11:01.44 | tainted- | so what's your argument |
| 11:01.52 | tainted- | them supportin *nix is a courtesy |
| 11:02.14 | tainted- | not an obligation |
| 11:02.27 | *** join/#asterisk riksta (~rick@81-178-209-106.dsl.pipex.com) |
| 11:02.33 | TUplink | will the diallogic work with zaptel? |
| 11:02.46 | TUplink | Non-Zaptel & Non-Dialogic Hardware |
| 11:02.51 | tainted- | try it |
| 11:02.53 | tainted- | wouldn't hurt |
| 11:03.03 | TUplink | is another section in hardware for asterisk |
| 11:03.15 | TUplink | how do i know if it dose work? |
| 11:03.29 | TUplink | i havent got my fxo cards yet |
| 11:04.05 | tainted- | google for "site:lists.digium.com dialogic" |
| 11:05.33 | jonathh | random question #2 any ways to connect a mobile to asterisk? |
| 11:06.55 | tainted- | define mobile |
| 11:07.10 | jonathh | well i for example have a sharp gx30 |
| 11:07.24 | jonathh | but even a mobile module.. so i can receieve and route calls to/from a sim? |
| 11:07.41 | tainted- | u'd need a gsm gateway |
| 11:08.03 | jonathh | are there interfaces with existing mobile out there? |
| 11:08.15 | tainted- | yes |
| 11:08.18 | tainted- | hardware interfaces |
| 11:08.21 | tainted- | called gsm gateways |
| 11:08.26 | jonathh | ah right :) |
| 11:08.30 | jonathh | googling now |
| 11:08.39 | tainted- | http://www.phonelabs.com/prd05.asp |
| 11:09.31 | jonathh | good man. |
| 11:09.35 | jonathh | that looks brill |
| 11:10.09 | *** join/#asterisk MrEntropy (~entropy@ppp38-183.lns1.adl1.internode.on.net) |
| 11:10.10 | MrEntropy | yo |
| 11:10.30 | tainted- | brill is what whales eat |
| 11:11.04 | tainted- | MrEntropy! |
| 11:11.09 | jonathh | isn't that krill? |
| 11:11.18 | MrEntropy | tainted-: hey there |
| 11:11.20 | *** join/#asterisk gres (~serg@81.222.48.242) |
| 11:11.41 | tainted- | oh yea |
| 11:11.49 | jonathh | :) |
| 11:11.49 | tainted- | i swear brill is seafood related too |
| 11:11.55 | jonathh | it is |
| 11:11.56 | jonathh | :) |
| 11:11.59 | jonathh | i just checked |
| 11:12.06 | jonathh | flat fish |
| 11:12.22 | tainted- | uwin |
| 11:12.25 | tainted- | buy me a dock-n-talk |
| 11:12.35 | jonathh | i win.. you buy ME one |
| 11:12.45 | tainted- | that's not brill |
| 11:12.51 | tainted- | brill is u buying me one |
| 11:12.54 | jonathh | obvious question is.. this can connect up to asterisk? |
| 11:13.15 | tainted- | yea converts gsm to analog rj45 |
| 11:13.18 | *** join/#asterisk RoyK (~roy@80.239.107.80) |
| 11:13.20 | jonathh | cool |
| 11:13.37 | jonathh | the iceing would be.. if it could handle more than one mobile at a time! |
| 11:14.59 | tainted- | brill icing |
| 11:15.08 | jonathh | krill icing |
| 11:15.21 | tainted- | touche |
| 11:15.40 | tainted- | they make gsm gateways in rackmounts |
| 11:15.47 | *** join/#asterisk Kal_Zakath (~Kal_Zakat@213.219.186.22.adslpower.by.edpnet.be) |
| 11:15.48 | jonathh | i can only conclude.. you too are bored out of your skull at work? |
| 11:15.50 | Kal_Zakath | hi |
| 11:16.04 | tainted- | Kal_Zakath! |
| 11:16.14 | Kal_Zakath | huh ? :) |
| 11:16.25 | tainted- | jonathh i'm procrastinating |
| 11:16.39 | jonathh | what ar e you meant to be doing? |
| 11:17.09 | tainted- | i was meant for sex and reproduction |
| 11:17.14 | Kal_Zakath | I'd like to send incoming call to the voice mail of the called user when I got this : Apr 21 13:04:39 NOTICE[6289]: app_dial.c:759 dial_exec: Unable to create channel of type 'SIP' |
| 11:17.18 | jonathh | okkk... |
| 11:17.21 | tainted- | but i'm should finish coding some stuff |
| 11:17.24 | Kal_Zakath | how can I do that ? |
| 11:17.40 | jonathh | might need a goto.. instead of a dial? |
| 11:17.46 | tainted- | Kal_Zakath what is your dial string |
| 11:18.35 | Kal_Zakath | someting like exten => s,2,Dial(SIP/myuser,20,r) |
| 11:19.12 | Kal_Zakath | works fine when the softphone of the user is registered to asterisk |
| 11:19.17 | tainted- | i'm guessing it's dial string syntax error |
| 11:19.59 | Kal_Zakath | but when the softphone is not open so not registered asterisk throw imediately the incoming call away, giving me this message |
| 11:20.37 | tainted- | paste your dialplan to pastebin.ca |
| 11:20.37 | MrEntropy | let's say i have two asterisk servers, if i get server #1(foo) to dial server #2(bar) using iax, does the entry on server #2 in iax.conf have to be [foo]? |
| 11:21.04 | Kal_Zakath | tainted-: ok, i'll paste the specific context |
| 11:21.09 | tainted- | MrEntropy register [bar] on [foo] and [foo] on bar |
| 11:21.35 | MrEntropy | tainted-: but register is only for dynamic ips |
| 11:21.36 | tainted- | Kal_Zakath does removing qualify = yes change anything? |
| 11:21.37 | *** join/#asterisk jabbzy (~dygup@noiseboys.force9.co.uk) |
| 11:21.57 | tainted- | MrEntropy ?? |
| 11:22.02 | Kal_Zakath | http://pastebin.ca/9989 |
| 11:22.12 | MrEntropy | tainted-: you meant the register=> command, no? |
| 11:22.24 | pigpen | hey...does any of the gentoo ebuilds for * do the spandsp patches to *? |
| 11:22.27 | MrEntropy | tainted-: you meant the register=> command, no? |
| 11:22.33 | MrEntropy | shit, sorry for dupe |
| 11:22.46 | tainted- | MrEntropy yea |
| 11:23.04 | MrEntropy | tainted-: my question was specific to the fact that, do the names in the square brackets have to match the name of the server? |
| 11:23.19 | Nuttah | anyone here use 1899.c0.uk as their voip carrier? |
| 11:23.24 | tainted- | MrEntropy just do IP |
| 11:23.38 | RoyK | http://pastebin.ca/10000 |
| 11:23.43 | MrEntropy | tainted-: in the square brackets? |
| 11:23.47 | Kal_Zakath | tainted-: nope, changing qualify value doesn't help |
| 11:25.00 | tainted- | register => username:password@fooIP:port/exten |
| 11:25.19 | tainted- | that would be for bar |
| 11:25.32 | MrEntropy | tainted-: the register => is only for dynamic ips though |
| 11:25.49 | tainted- | dynamic IPs? |
| 11:25.55 | tainted- | are foo and bar on static IPs? |
| 11:25.59 | MrEntropy | yes |
| 11:26.06 | tainted- | then just put their static IPs in there |
| 11:26.14 | MrEntropy | but why would i register? |
| 11:26.22 | tainted- | why would register be restricted to dynamic IPs |
| 11:26.43 | tainted- | to give the servers access to one another? |
| 11:26.44 | MrEntropy | its not, but the manual says on static ips it's superfluous |
| 11:27.23 | Nuttah | its not needed with static ips |
| 11:27.23 | tainted- | how is it not needed? |
| 11:27.23 | tainted- | curious |
| 11:27.26 | Nuttah | if either end is dynamic.. it is. |
| 11:27.46 | MrEntropy | because on static ip's, appropriate peer entries in iax.conf with ip's are enough |
| 11:28.25 | Nuttah | or hostnames |
| 11:28.27 | MrEntropy | something's wrong with one of my servers though, and i'm trying to figure out whether i have a match wrong |
| 11:28.51 | MrEntropy | so i'm asking whether on bar, the entry needs to be [foo] because [from-foo] won't match? |
| 11:29.31 | Nuttah | incoming i'm not sure.. but outgoing i dont believe it needs to match |
| 11:29.55 | MrEntropy | ok, now where do i set the name of the asterisk server? |
| 11:30.35 | Nuttah | you mean the hostname? |
| 11:30.42 | MrEntropy | so it just uses hostname? |
| 11:30.46 | Kal_Zakath | tainted-: nevermind, i'm a stupid moron, I messed up with variable |
| 11:30.51 | Kal_Zakath | thanks anyway |
| 11:30.53 | Kal_Zakath | :D |
| 11:31.06 | Kal_Zakath | works fine now |
| 11:31.15 | Nuttah | hostname is set outside if asterisk |
| 11:31.24 | Nuttah | hostname is set outside asterisk |
| 11:31.35 | MrEntropy | oh i know...i just wasn't aware it uses hostname |
| 11:32.33 | Nuttah | it can use either tbh |
| 11:32.40 | Nuttah | IP should work |
| 11:33.55 | tainted- | Kal_Zakath what was it? |
| 11:34.42 | Kal_Zakath | tainted-: stupid mistakes with variables because of a stupid copy/paste in my extensions.conf :D |
| 11:35.05 | tainted- | oh |
| 11:35.39 | Kal_Zakath | could be qualified as a stupid n00b mistake :D |
| 11:35.51 | tainted- | typos get everyone |
| 11:36.16 | MrEntropy | wtf? seriously...i have them match now, foo sends a call to bar, bar recieves it(because i ngrep the port) but there is no display in the asterisk console, not even in iax2 debug! |
| 11:37.09 | MrEntropy | packets arrive, but asterisk does nothing, any idea? |
| 11:37.12 | tainted- | did u make sure u 'relaod' |
| 11:37.17 | tainted- | 'reload' |
| 11:37.33 | MrEntropy | yes |
| 11:37.45 | tainted- | what do u see when do u iax2 show peers |
| 11:38.17 | MrEntropy | on bar, i see foo(unmonitored) |
| 11:38.42 | tainted- | how are u handling the call on bar |
| 11:39.17 | MrEntropy | gets dumped to context 'to-pstn', where the extention is dialed out a zap channel |
| 11:39.32 | MrEntropy | but i don't think i'm that far in...asterisk fails to respond to the packet |
| 11:39.48 | MrEntropy | would i be correct in saying that i should at least get a fat REJECTED or something? |
| 11:39.52 | tainted- | what does foo say |
| 11:40.07 | Kal_Zakath | well thanks guy, keep up the good work |
| 11:40.10 | Kal_Zakath | cya |
| 11:40.14 | *** part/#asterisk Kal_Zakath (~Kal_Zakat@213.219.186.22.adslpower.by.edpnet.be) |
| 11:40.56 | MrEntropy | Executing Dial("SIP/asterisk-896f", "IAX2/bar/82616925") in new stack |
| 11:41.14 | MrEntropy | and in iax2 debug on foo i get stacks of output |
| 11:42.45 | tainted- | try just NoOp(some stuff) in [to-pstn] on bar |
| 11:43.00 | tainted- | maybe u have something crapping out in dialplan |
| 11:43.23 | MrEntropy | would that really prohibit iax2 debug output? |
| 11:44.55 | tainted- | dunno |
| 11:45.18 | tainted- | you're practically there though |
| 11:46.00 | MrEntropy | got it... |
| 11:46.03 | MrEntropy | port |
| 11:46.13 | tainted- | nice |
| 11:46.35 | MrEntropy | the manual said default was 5036 so i changed it to that, but for some reason it has to be 4569 |
| 11:47.04 | MrEntropy | i need to stop listening to that manual so often =) |
| 11:47.08 | tainted- | i have 5036 |
| 11:47.16 | tainted- | so u put 4569 on both side? |
| 11:47.21 | tainted- | s |
| 11:47.33 | tainted- | or have u tested foo->yet |
| 11:47.51 | MrEntropy | no! and that's the thing, one side is foo is 5036 and bar has to be 4569 for obscure reasons |
| 11:48.15 | tainted- | where are u setting this |
| 11:48.21 | MrEntropy | iax.conf |
| 11:48.27 | Nuttah | odd... both my sides use udp 4569 |
| 11:48.34 | Nuttah | as default |
| 11:48.36 | tainted- | [general] or in the [bar] context |
| 11:49.42 | MrEntropy | [general] |
| 11:49.51 | MrEntropy | general on both machines |
| 11:50.35 | MrEntropy | maybe iax is 4569 and iax2 is 5063? |
| 11:50.50 | *** part/#asterisk Kumbang (~ecvs@167.205.24.4) |
| 11:50.56 | MrEntropy | just a thought |
| 11:51.09 | *** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net) |
| 11:53.14 | Nuttah | other way around |
| 11:53.33 | Nuttah | iax2 uses 4569 iax uses 5036 |
| 11:54.12 | Nuttah | http://www.voip-info.org/wiki-IAX |
| 11:55.38 | zoa | dontus eiax |
| 11:55.44 | zoa | its mot working anyway |
| 11:57.20 | Nuttah | sees fine to me |
| 11:57.23 | Nuttah | seems |
| 11:57.44 | zoa | dont use iax |
| 11:57.48 | zoa | use iax2 i meant |
| 11:57.48 | *** join/#asterisk peter222 (peter222@dsl-202-173-142-98.sa.westnet.com.au) |
| 11:59.38 | Nuttah | ah :P |
| 12:00.53 | *** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 12:02.17 | *** join/#asterisk darby_t (~tom@host-ip226-209.crowley.pl) |
| 12:07.28 | *** join/#asterisk kanzure (~bbishop@cpe-66-68-141-11.austin.res.rr.com) |
| 12:08.43 | Zeeek | heh |
| 12:10.05 | *** join/#asterisk n4y (~frodo7@host-ip226-209.crowley.pl) |
| 12:21.44 | *** join/#asterisk Fabe (~spamhere@pD95B0BF7.dip0.t-ipconnect.de) |
| 12:22.14 | MrEntropy | thanks for all the help guys |
| 12:22.19 | MrEntropy | tata! |
| 12:22.53 | Zeeek | everything is everything |
| 12:24.08 | kanzure | Zeeek, does that make me everyone? |
| 12:26.41 | *** join/#asterisk Ahrimanes (~michael@gw-ext.catpipe.dk) |
| 12:28.02 | newl | only if you're making yourself. |
| 12:28.04 | *** join/#asterisk ckruetze (HydraIRC@cpc3-cmbg7-5-0-cust100.cmbg.cable.ntl.com) |
| 12:28.32 | kanzure | newl: normal routine? configure, make, make-install right? |
| 12:28.51 | *** join/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net) |
| 12:29.18 | newl | kanzure: For your default location, otherwise you'll need to specify --prefix. :) |
| 12:29.20 | Zeeek | kanzure it depends on your karma |
| 12:29.24 | oden | newl: new kiax package in mandriva works on x86_64 too :-) |
| 12:29.34 | newl | oden: cool beans :) |
| 12:31.25 | jabbzy | appolgies for the long windedness of this. i want to make some extensions not dialble from others, yet give them a "public" extention (ring group) to dial. say i have 2 user groups regulars and medic. regulars can't dial a medics extention directly, but they can dial medics hunt group. will the hunt work? |
| 12:31.34 | Zeeek | iax2 question. I have a phone stiing here connecting to a server on a different network. For some reason, it doesn't want to use 4569 and I can't figure out why? |
| 12:32.11 | Zeeek | jabbzy the context is what is protected, not the phone |
| 12:32.31 | Zeeek | so once in a context, put the hunt group code in THAT context |
| 12:33.03 | Zeeek | for example people in the [unclean] context dial 6969 for the nunt group |
| 12:33.16 | Zeeek | s/nunt/hunt/ |
| 12:35.02 | jabbzy | thanks zeek.. i thought that was as it worked... its kind of like phone natting. what was the s/hunt/hunt/ for? |
| 12:35.15 | Zeeek | I misspelled hunt |
| 12:35.39 | jabbzy | :) |
| 12:35.57 | newl | heh wicked..called from my GS->*->fwd to fwd->kiax and there was a 10 second lag |
| 12:38.13 | jabbzy | what i need is an asterisk visualiser! something that can read the configs and display what goes where... |
| 12:38.43 | *** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com) |
| 12:40.06 | *** join/#asterisk _SMP_ (~SMP@pandora.burned.net) |
| 12:42.07 | Zeeek | are you using @hole or something? |
| 12:43.22 | jabbzy | right so you set up contexts like [medic] [regular] [swichboard] [external] and then do an include=> [from-regular] [from-external-allow] to the contexts in extensions.conf? |
| 12:46.13 | Zeeek | look at it this way, whatever access the system comes in to a context. Put restricted services and numbers in separate contexts. Include those restricted service contexts as needed in the less privileged ones |
| 12:46.42 | Zeeek | there are a lot fo pages on the wiki about dialplans including some thoughts on logical ways to organize |
| 12:46.45 | Zeeek | The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. |
| 12:46.45 | Zeeek | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
| 12:46.47 | Zeeek | oops |
| 12:46.49 | kajtzu | hmm |
| 12:47.13 | Zeeek | thought I had them here but I don't |
| 12:47.30 | kajtzu | is there any way to figure out why some calls seem hung (show sip channels shows 5 active calls of which 4 are from the same peer but the peer hasn't existed for 3-4 hours ) |
| 12:47.35 | *** join/#asterisk styx2005 (~styx2005@a-line138.supra.net) |
| 12:48.02 | Zeeek | aren't those calls just info checks? |
| 12:48.13 | Zeeek | like register, qualify... |
| 12:48.18 | styx2005 | can anybody tell me the default login for amp in asterisk@home? |
| 12:48.33 | Zeeek | see the mailing list, it comes up allthe time |
| 12:48.40 | jabbzy | thanks again Zeeek, i have done simpler dial plans before, but this one is getting complicated, just needs some good planning and understanding. |
| 12:48.46 | jabbzy | thanks again Zeeek, i have done simpler dial plans before, but this one is getting complicated, just needs some good planning and understanding. |
| 12:48.52 | jabbzy | erk sorry. |
| 12:48.58 | Zeeek | jabbzy it's headache material all reet |
| 12:49.26 | kajtzu | Zeeek: how do I find out if it's a info message? sip show channels doesn't tell me that :) |
| 12:49.29 | Zeeek | the other day, I had one of our incoming lines looping on calls. Removed a line and had a bad goto |
| 12:49.55 | Zeeek | kajtzu dunno, sorry, but that's what I think I've seen on my sys |
| 12:50.10 | kajtzu | doing sip show channel ... it says SIP Call |
| 12:50.47 | Zeeek | format unknown? |
| 12:50.58 | Zeeek | User/ANR = none? |
| 12:51.52 | kajtzu | <PROTECTED> |
| 12:52.00 | kajtzu | it looks just like a regular call |
| 12:52.34 | Zeeek | no username? |
| 12:52.44 | kajtzu | username is legal and what it's supposed to be |
| 12:52.56 | kajtzu | lastmessage is Tx: ACK |
| 12:53.00 | TheEmperor | hey guys |
| 12:53.11 | kanzure | hello |
| 12:53.18 | Zeeek | are these calls that use reinvite and leave your * box? |
| 12:53.33 | kajtzu | hmmm |
| 12:53.54 | *** join/#asterisk Romik (~romik@router-net.ser.netvision.net.il) |
| 12:54.01 | *** part/#asterisk styx2005 (~styx2005@a-line138.supra.net) |
| 12:54.42 | kajtzu | reinvite is set to no |
| 12:54.44 | kajtzu | for that user |
| 12:54.59 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 12:59.02 | *** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc) |
| 12:59.57 | saabluvr | I'm having trouble with spandsp / rxfax : asterisk immediately hangs up after accepting the call |
| 13:00.00 | saabluvr | <PROTECTED> |
| 13:00.03 | saabluvr | *CLI> -- Executing RxFAX("Zap/1-1", "/home/master/testfax.tif") in new stack |
| 13:00.06 | saabluvr | <PROTECTED> |
| 13:00.08 | saabluvr | <PROTECTED> |
| 13:00.23 | *** join/#asterisk sambal (~sambal@gateway.office.flatbox.nl) |
| 13:01.02 | saabluvr | My system is an up-to-date debian sarge, and i let asterisk compile by executing the script in bristuff |
| 13:01.31 | saabluvr | by itself asterisk runs fine , just not the fax app |
| 13:02.23 | *** join/#asterisk satlink (satlink@66.178.97.50) |
| 13:02.42 | saabluvr | on a machine running the Digium E1 card all works fine , but not on the machine running zaphfc |
| 13:03.10 | saabluvr | what could cause the hangup ? Any ideas ? |
| 13:03.20 | Essobi | saabluvr /home/master exist and has permissions to be seen/wrote to by the user you're running asterisk as? |
| 13:03.51 | saabluvr | I am running asterisk as root :) |
| 13:03.58 | Essobi | Okay.. |
| 13:04.02 | Essobi | saabluvr /home/master exist and has permissions to be seen/wrote to by the user you're running asterisk as? |
| 13:04.06 | Essobi | :) |
| 13:05.32 | saabluvr | I compiled and undid everything at least 3 times now :( the problem stays |
| 13:05.49 | satlink | Hey! Problem: Two SIP clients with different IP's+NAT can communicate audio both ways with for example Voicemail, but not with each other.. |
| 13:05.51 | Essobi | ls -ld /home/master? |
| 13:05.59 | satlink | Any clues? keywords where to look in the lists? |
| 13:06.18 | saabluvr | drwxr-xr-x 2 master master 4096 2005-04-21 08:52 /home/master/ |
| 13:06.30 | Essobi | satlink easy enough, they need to be bridged on your * box, as the RTP isn't double nat aware. |
| 13:06.42 | Essobi | satlink Or they just have a netsplit between them.. is it transient? |
| 13:07.09 | satlink | ok. thanks, still your language is a bit cryptic for me:) |
| 13:07.13 | Essobi | saabluvr No idea.. You borked something.. spandsp has always just worked for me. Debian sarge too. |
| 13:07.22 | Essobi | satlink Learn VoIP. :) |
| 13:07.29 | Essobi | RTP = the voice stream. |
| 13:07.30 | satlink | working on it.. |
| 13:07.44 | satlink | got that, but what do you mean with transient? |
| 13:07.49 | Essobi | Netsplit. |
| 13:08.09 | Essobi | Control leg that makes the phone does like this.. |
| 13:08.22 | Essobi | Site A <-> * <-> Site B |
| 13:08.39 | Essobi | RTP (voice) can go that way too |
| 13:08.41 | Essobi | OR.. |
| 13:08.52 | Essobi | It'll go A <-> B if it thinks it can. |
| 13:09.23 | satlink | ok. and you think my clients believe they can go directly, but thay can not because of the nat's? |
| 13:09.30 | Essobi | and bypass *.. A can talk to *, B can talk to *, but A can't talk to B directly due to firewalls, routes down, bad routes, etc. |
| 13:09.42 | Essobi | Sounds like it. |
| 13:09.58 | satlink | how do I force them to speak through * |
| 13:10.01 | Essobi | Disable reinvite on both of the sip peers in /etc/asterisk.conf. |
| 13:10.08 | Essobi | I was already typing it. ;) |
| 13:10.20 | Essobi | err |
| 13:10.28 | Essobi | <PROTECTED> |
| 13:10.33 | satlink | ok, thanks alot, as said, new to the game.. |
| 13:10.45 | bjohnson | satlink: as a quick test, since they can both speak to asterisk .. add canreinvite=no to both devices sip.conf entries |
| 13:11.09 | TheEmperor | everytime i call someone, they suffer from hearing themselves after they have spoken, any ideas anyone? |
| 13:11.13 | Essobi | satlink Game? Pfft, I wish this thinx had a ps2 controller.. I'd pwn. |
| 13:11.20 | satlink | I was not familiar with the reinvite=no in asterisk.conf, but i new about it in sip.conf, even though i did not knwo the full meaning of it.. |
| 13:11.22 | bjohnson | TheEmperor: it's called echo |
| 13:11.36 | satlink | hehe |
| 13:11.43 | bjohnson | TheEmperor: usually turning down the output gain from the person speaking helps |
| 13:11.49 | TheEmperor | bjohnson: how can i rectify this? |
| 13:11.50 | TheEmperor | ok |
| 13:12.00 | TheEmperor | thing is i am using * to call their mobile |
| 13:12.02 | *** join/#asterisk jalsot_ (~chatzilla@217.116.36.22) |
| 13:12.05 | satlink | Essobi I will try.. I'll be back! thanks |
| 13:12.08 | Essobi | Yea, he's right.. canreinvite |
| 13:12.16 | Manipura | satlink, I thought it was canreinvite |
| 13:12.19 | bjohnson | TheEmperor: turn down the output gain on your fxo |
| 13:12.28 | Essobi | Oh, and boys and girls.. www.voip-info.org is your friend.. love it, google it. |
| 13:12.34 | bjohnson | ~docs |
| 13:12.36 | 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 |
| 13:12.36 | TheEmperor | bjohnson: i'm using firefly.. |
| 13:12.52 | Essobi | Wooooo, BigJohnson with the docs loving. |
| 13:13.06 | Essobi | ~seen Oprah's ass, but I hear it's quite nice. |
| 13:13.14 | jbot | Essobi: i haven't seen 'oprah's ass, but i hear it's quite nice.' |
| 13:13.14 | bjohnson | TheEmperor: I don't use firefly so don't know it's controls. However, I was referring to asterisk |
| 13:13.24 | Essobi | Firefly made my box lock up last time I installed it. |
| 13:13.28 | bjohnson | hehe .. on a whim I registered bigjohnson.ca yesterday |
| 13:13.34 | bjohnson | doesn't point to anything yet though |
| 13:13.36 | Essobi | mehe |
| 13:13.41 | TheEmperor | bjohnson: how would i turn down the fxo gain on my asterisk box? |
| 13:13.42 | Essobi | He said point. |
| 13:13.54 | satlink | sorry bothering you guys..but thanks again.. |
| 13:13.55 | bjohnson | TheEmperor: how is asterisk calling the cell phone? |
| 13:14.04 | Essobi | TheEmperor You didn't get the FXO gain 1.0 knob installed?!?!! |
| 13:14.15 | TheEmperor | Essobi: haha |
| 13:14.18 | *** part/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 13:14.18 | Essobi | satlink Ehh? |
| 13:14.22 | bjohnson | TheEmperor: you're calling out of a fxo right? |
| 13:14.30 | TheEmperor | bjohnson:no, using nufone |
| 13:14.44 | bjohnson | hmm .. that's different then |
| 13:14.51 | TheEmperor | bjohnson: any ideas? |
| 13:14.55 | bjohnson | echo isn't usually a problem with voip providers |
| 13:14.58 | Essobi | satlink You all straightened out? |
| 13:15.06 | bjohnson | echo is usually a problem with fxo connections |
| 13:15.13 | saabluvr | Essobi: did u use spandsp with zaphfc ? |
| 13:15.16 | Essobi | Usually. |
| 13:15.16 | TheEmperor | bjohnson: i see |
| 13:15.29 | Essobi | saabluvr I don't use any hardware. I'm all net, baby. |
| 13:15.52 | bjohnson | TheEmperor: try calling some other people and see if you can determine a pattern to the problem ie .. happens with some cell phones but never a problem with land line phones |
| 13:16.01 | Essobi | Seriously.. No zap gear here.. |
| 13:16.08 | TheEmperor | bjohnson: good idea, will try that |
| 13:16.13 | bjohnson | TheEmperor: sounds like a problem to report to nufone |
| 13:16.15 | Essobi | I know plenty of people in here have used it.. |
| 13:16.29 | Essobi | saabluvr try posting to -user and see if anyone else has the same problems.. |
| 13:16.30 | *** join/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 13:16.36 | TheEmperor | bjohnson: i tried with voipjet, seems to be the same problem.. |
| 13:16.37 | Essobi | What verbose and debug levels you got set? |
| 13:16.57 | bjohnson | TheEmperor: also maybe try a different softphone |
| 13:17.06 | bjohnson | TheEmperor: oh wait .. just had a thought |
| 13:17.13 | TheEmperor | ? |
| 13:17.18 | Essobi | saabluvr Try 5 or so on both and send the log of the console up to pastebin.ca right quick. |
| 13:17.19 | bjohnson | TheEmperor: I don't use softphones so forgot this item |
| 13:17.33 | bjohnson | TheEmperor: try turning down the mic volume on the softphone machine |
| 13:17.33 | Essobi | Heh. |
| 13:17.44 | TheEmperor | bjohnson:ah.. that could be an idea... |
| 13:17.52 | Essobi | Or the speakers.. if you're using spkrs instead of headphones. |
| 13:18.07 | *** join/#asterisk km- (pgrace@67.105.178.130) |
| 13:18.15 | Essobi | Or turn it up, if you love Jimi Hendrix. :) |
| 13:18.19 | Gand_DJ | what's his issue? |
| 13:18.24 | bjohnson | echo |
| 13:18.30 | bjohnson | echo through a voip provider |
| 13:18.32 | km- | howdy! |
| 13:18.40 | bjohnson | from a softphone to a cell phone |
| 13:18.49 | Essobi | NEEEEEEER NO NEEER NEEEER.... NEEERNEEERNEEEEER WOA, YEA.. PURPLE HAZE ALL IN MY BRIAN... |
| 13:18.59 | Gand_DJ | I know with 1 of my softphones, I sound kinda choppy using gsm.... but using another softphone on gsm is fine (but quieter) running via * |
| 13:19.05 | bjohnson | I think you spelt NEEEEEEER wrong |
| 13:19.27 | *** join/#asterisk SuPrSluG_ (~SuPrSluG@pool-129-44-142-202.buff.east.verizon.net) |
| 13:19.31 | TheEmperor | this is strange |
| 13:19.35 | km- | 7940 has the same grade speakerphone as the 7960, right? |
| 13:19.45 | TheEmperor | i called a landline and they can hear me ok but i get echo on my headset now |
| 13:20.21 | bjohnson | try playing with volumes .. keep calling the girlfriend and tell her you just want to keep hearing her voice |
| 13:20.28 | Essobi | km- I think the 40 is half duplex |
| 13:20.46 | Essobi | Or maybe that was the one below the 40.. |
| 13:20.58 | Essobi | I can't remember.. they're all shite save the 60.. |
| 13:21.02 | Essobi | BUY THE 60s! |
| 13:21.22 | km- | yeah, I'm thinking of going with the 60 |
| 13:21.27 | km- | anyone know if there's a sip load for the 7970? |
| 13:21.30 | TheEmperor | Gand_DJ: what softphone do you use? |
| 13:21.58 | Gand_DJ | I use Eyebeam mainly, and sometimes Firefly |
| 13:21.59 | TheEmperor | i am using firefly since it uses IAX2, what other softphones use iax2? |
| 13:22.26 | TheEmperor | Gand_DJ: Yeah, sometimes FireFly is working well, other times its not so good... |
| 13:22.38 | *** join/#asterisk jtar (~zic@81-178-54-38.dsl.pipex.com) |
| 13:23.27 | Gand_DJ | For some reason when I did a test call to someone in another province (who is setup as an extension in my *). he said I sounded kinda choppy / digital using eyebeam. |
| 13:23.47 | Gand_DJ | When I retried with firefly (which he used also) on gsm, it was fine.. but audio was quieter on his end |
| 13:24.05 | Essobi | I don't use EB, just the phone. |
| 13:24.14 | *** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net) |
| 13:24.28 | Essobi | And mostly that's when I'm too lazy to turn around and grab one of the 5 phones mounted on my wall. :) |
| 13:24.29 | Gand_DJ | I like eyebeam because it allows 6 channels at once, conferencing, transfers, parking |
| 13:24.34 | bjohnson | TheEmperor: I think diax is another windoze iax softphone |
| 13:24.36 | TheEmperor | Gand_DJ:So i'm not sure what to do with this echo |
| 13:24.48 | Essobi | :) |
| 13:25.21 | *** join/#asterisk DrWho17 (~MIKE@mike-new.tc3net.com) |
| 13:25.36 | bjohnson | TheEmperor: make some money from it |
| 13:25.43 | TheEmperor | ? |
| 13:25.49 | Gand_DJ | maybe adjusting jitter will fix echo? not sure what settings firefly has without firing it up..lol |
| 13:25.59 | bjohnson | TheEmperor: sell people the chance to "call the Grand Canyon and hear their own echo" |
| 13:26.05 | TheEmperor | hahah |
| 13:26.25 | Gand_DJ | I'm still browsing for a nice provider for outbound, and maybe inbound, for my * box |
| 13:26.27 | *** join/#asterisk pjm_uk (~pjm_uk@cpc1-pool3-3-0-cust116.sot3.cable.ntl.com) |
| 13:26.27 | bjohnson | $1 a call .. 10 seconds each |
| 13:26.53 | Gand_DJ | Been reading mixed info on alot of providers out there..lol |
| 13:26.55 | satlink | Essobi: canreinvite=no did the trick... |
| 13:26.59 | Gand_DJ | expecially Broadvoice |
| 13:27.09 | Essobi | satlink Woohoo. |
| 13:27.14 | TheEmperor | i am wondering if i should modify the jitterbuffer settings in my * box? |
| 13:27.17 | Essobi | Yea.. NAT is 3V1L.. |
| 13:27.19 | pjm_uk | afternoon all, could someone recommend a web based manager for * that could allow user account management? I've tried AMP, but it doesnt allow the easy importing of existing * users... |
| 13:28.18 | bjohnson | pjm_uk: but how often would you do that? |
| 13:28.26 | bjohnson | I think just once |
| 13:29.53 | bjohnson | pjm_uk: from what I can understand .. a web interface doesn't yet exist that provides enough flexibility for people to really use |
| 13:30.02 | Essobi | bjohnson Anope. |
| 13:30.12 | Essobi | I was thinking of making a stupid one.. |
| 13:30.16 | bjohnson | mainly cause the idea that a gui should make it simpler .. and being flexible is not simple |
| 13:30.55 | Essobi | Nah, screw simple.. I want dialplan editing in a web interface with history, |
| 13:31.21 | Essobi | and not having to renumber all my damn entries when I want to add/delete something in the middle. |
| 13:31.58 | Essobi | And that's good enough for me. |
| 13:32.09 | Essobi | and ACLs for contexts would be nice too. |
| 13:32.32 | Essobi | So someone can entriely piss up everything.. just their context. :)\ |
| 13:32.52 | pjm_uk | bjohnson: ok thanks for that ... perhaps I need to make one in vb then... its basically for admin people to reset passwords and add new extensions etc |
| 13:33.20 | *** part/#asterisk saabluvr (master@keeper.nc-ks.de) |
| 13:33.36 | pjm_uk | i guess the answer is to use the dial plan and extensions.conf from a mysql, then odbc into that to change parms etc |
| 13:34.16 | *** join/#asterisk tzafrir_laptop (~tzafrir@62.90.10.53) |
| 13:34.45 | *** join/#asterisk basta (~kqj@62-101-126-233.fastres.net) |
| 13:34.57 | TheEmperor | another weird issue i always face is that asterisk seems to say that an extension has hanged up when i actually hasn't and the call is still in progress |
| 13:35.05 | TheEmperor | like when i use a provider |
| 13:35.29 | Essobi | pjm_uk Just update the database ACIDly or the filesystem atomicly |
| 13:35.54 | basta | what's the right configuration for having DTMF work with sip on cisco phones ? |
| 13:35.56 | TheEmperor | this is bad, how would i bill my customers correctly?? |
| 13:36.00 | Essobi | TheEmperor You got those silly signal detection routines compiled in |
| 13:36.11 | *** join/#asterisk iq (~iq@63-230-44-31.omah.qwest.net) |
| 13:36.13 | TheEmperor | Essobi: how would i remove them? |
| 13:36.26 | Essobi | Or maybe not.. |
| 13:36.34 | dmccollum | Has anyone successfully got AMPortal to work with realtime? |
| 13:36.37 | Essobi | elaborate on the situation while I dig up the code.. |
| 13:36.48 | *** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk) |
| 13:36.48 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 13:36.49 | Essobi | AMP is shite my friend, and no they havn't. |
| 13:36.55 | *** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:36.55 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:37.06 | Essobi | Howdy anthm |
| 13:37.12 | TheEmperor | well, when i dial out on say livevoip or other providers, it shows on my command line that the extension has hung up |
| 13:37.20 | TheEmperor | but the call is still in progress |
| 13:37.27 | TheEmperor | so that's going to be bad for billing |
| 13:37.30 | Gand_DJ | Hrm, would having VAD enabled on a softphone cause me to sound choppy? |
| 13:37.37 | Essobi | show channels? and you checked CDRs for actual duration? What protocols. |
| 13:37.43 | Essobi | Gand_DJ It can. |
| 13:37.44 | TheEmperor | iax2 |
| 13:37.47 | anthm | hi |
| 13:38.10 | *** join/#asterisk tessier (~treed@210.245.103.132) |
| 13:38.14 | Essobi | TheEmperor You're reinviting your self out of the stream. IAX has no way of knowing how long a call is unless it stays in the stream. |
| 13:38.25 | Essobi | Hence * can't tell you the real duration. |
| 13:38.27 | TheEmperor | Essobi: I see, how would i rectify this? |
| 13:38.53 | Essobi | I don't use IAX.. I assume it's canreinvite like in sip, but go check the iax.conf.sample |
| 13:38.57 | Essobi | or voip-info.org |
| 13:39.18 | Essobi | Cause I can be a stupid S.O.B. sometimes. :) |
| 13:39.50 | TheEmperor | :) |
| 13:41.50 | *** join/#asterisk ToyMan (~konversat@204-8-82-238.webjogger.net) |
| 13:42.21 | *** join/#asterisk moy (~kvirc@201.135.105.124) |
| 13:43.23 | *** join/#asterisk SuPrSluG_ (~SuPrSluG@pool-129-44-142-202.buff.east.verizon.net) |
| 13:43.37 | km- | you can make it so the call wont transfer off |
| 13:43.53 | km- | the particular config option eludes me at the moment though |
| 13:44.18 | Essobi | So anyone going to be in Louisville, Kentucky this weekend? :) |
| 13:44.31 | km- | essobi: anything fun going on there? |
| 13:44.48 | km- | It's a 14 hour drive for me to get there :P |
| 13:45.00 | Luhiwu | anyone knows if it is possible to log the termination reason in the cdr? |
| 13:45.10 | Essobi | http://www.thunderoverlouisville.org/theshow/ |
| 13:45.24 | Essobi | :) |
| 13:45.40 | Essobi | Only the biggest fireworks display in the USA. |
| 13:46.02 | Essobi | http://community.webshots.com/album/36441741xbAIZy |
| 13:46.07 | Essobi | There you go.. some good pics. |
| 13:46.26 | *** join/#asterisk dreamcode (~alone@81.181.199.33) |
| 13:46.35 | dreamcode | hello |
| 13:46.59 | km- | essobi: nice. |
| 13:47.30 | dreamcode | hello all |
| 13:47.57 | km- | will forcing chan_oss to noload stop MOH? |
| 13:48.41 | Essobi | Uhh.. I wouldn't think so. |
| 13:49.37 | *** join/#asterisk MeTaBSD (metabsd@your.axx.is.denied.ws) |
| 13:49.40 | MeTaBSD | hi :) |
| 13:49.50 | km- | hmm |
| 13:49.58 | km- | I think I have a stale channel |
| 13:52.01 | RoyK | km-: restart now usually helps |
| 13:52.25 | dreamcode | how many simultaneos channels works on asterisk on a PIII@500MHz with 196MB RAM ? |
| 13:52.49 | RoyK | dreamcode: how tall is a tree? |
| 13:53.05 | dreamcode | why ? |
| 13:53.25 | RoyK | dreamcode: what are you trying to do? using pots? isdn? sip? mgcp? transcoding anything? |
| 13:53.29 | RoyK | _what_? |
| 13:53.41 | km- | royk: haha, word.. |
| 13:53.51 | RoyK | you're asking what's like "how much memory does a server need for 10 users" |
| 13:53.56 | RoyK | which is - well - enough |
| 13:54.37 | dreamcode | i'm asky because sometime.. i have a call hang... meaninig... a call that doesn hangup.. about 9H |
| 13:54.37 | *** join/#asterisk _m_ (~m@fbta199.fbta.uni-karlsruhe.de) |
| 13:54.39 | RoyK | my exact, and correct, answer to your initial question is "probably quite a few, but not as many as a faster computer" |
| 13:55.04 | RoyK | dreamcode: using RTP? |
| 13:55.10 | dreamcode | yes |
| 13:55.31 | RoyK | forgot to set rtptimeout= in sip.conf? |
| 13:55.35 | RoyK | (if you use sip) |
| 13:55.39 | dreamcode | i have set it |
| 13:55.59 | km- | dreamcode: I have a P3 550mhz with 128mb of ram and I can handle two t1's worth |
| 13:56.12 | RoyK | km-: wot??? |
| 13:56.14 | Gand_DJ | Is there a way to setup the Digital Assistant / IVR so that after the menu is played, if nothing is pressed after xx seconds, a second menu is played stating nothing was pressed, and then repeats previous menu..... and after 3 menu repeats, if nothing is pressed, call is terminated |
| 13:56.21 | RoyK | km-: that's a lot... |
| 13:56.27 | km- | yeah, it can handle it pretty wel |
| 13:56.32 | km- | it's more or less just audio bridging |
| 13:56.36 | km- | when you're doing transcoding |
| 13:56.37 | RoyK | no echo cancellation, then... ? |
| 13:56.39 | km- | like sip and such |
| 13:56.42 | km- | nope, echo cancellation as well |
| 13:56.46 | RoyK | k |
| 13:56.50 | RoyK | not bad, really |
| 13:56.56 | RoyK | km-: and no memory leaks? |
| 13:57.01 | km- | if you had 20 users using gsm at once I'm sure the box would dog a lil |
| 13:57.07 | km- | lemme check.... |
| 13:57.11 | RoyK | I'm haunted by them |
| 13:57.33 | RoyK | we currently do ~30-50 g.729a transcodings |
| 13:57.38 | km- | 35megs of active cache with 5megs free ram on the box at the moment |
| 13:58.02 | km- | and there are 9 channels active at the moment |
| 13:58.04 | RoyK | strange. my asterisk processes tend to grow, eating more memory every day |
| 13:58.06 | km- | 10 |
| 13:58.10 | RoyK | km-: show uptime |
| 13:58.10 | km- | my phone just started ringing |
| 13:58.11 | km- | :) |
| 13:58.12 | SuPrSluG_ | anyone use LCDial |
| 13:58.13 | dreamcode | but is there any aplication that will hangup a call that last more than 1 hour ? |
| 13:58.32 | km- | one min |
| 13:58.41 | km- | sf01*CLI> show uptime |
| 13:58.41 | km- | System uptime: 20 hours, 41 minutes, 48 seconds |
| 13:58.42 | km- | Last reload: 19 hours, 40 minutes, 45 seconds |
| 13:58.53 | RoyK | dreamcode: AbsoluteTimeout |
| 13:59.19 | km- | so we're about 21 hours into the session and everything's hunky-dory so far |
| 13:59.32 | km- | this is a 100% production system at this point |
| 13:59.54 | km- | we are a real company doing real work using Asterisk, which I hear is hard to run into for case study purposes |
| 13:59.59 | bjohnson | Gand_DJ: yes |
| 14:00.03 | Gand_DJ | asterisk1*CLI> show uptime |
| 14:00.03 | Gand_DJ | System uptime: 1 week, 20 hours, 32 minutes, 22 seconds |
| 14:00.03 | Gand_DJ | Last reload: 18 hours, 15 minutes, 43 seconds |
| 14:00.06 | Gand_DJ | :) |
| 14:00.09 | bjohnson | Gand_DJ: you would do that with your dial plan |
| 14:00.09 | dreamcode | i had about 7 days * up.. and today.. just died on me.. :( |
| 14:00.11 | km- | nice uptime there :) |
| 14:00.28 | *** join/#asterisk MattH (~matth@noc-wireless.chilitech.net) |
| 14:00.30 | km- | I'm finding that most of my headaches now are with the NEC system convergence |
| 14:00.34 | RoyK | km- what? |
| 14:00.38 | MattH | Hi... how can I go about preventing someone from RECEIVING calls? |
| 14:00.55 | km- | royk: we have Asterisk acting as the primary pbx, and have an NEC attached to it off a te405 |
| 14:01.03 | RoyK | kk |
| 14:01.08 | km- | royk: we're transitioning off the NEC onto full VoIP |
| 14:01.09 | RoyK | MattH: see extensions.conf |
| 14:01.46 | bjohnson | MattH: prevent from receiving any calls? just don't have an exten to their phone |
| 14:02.09 | MattH | bjohnson: so just authenticate them but don't point the extension for the 'username' at the phone? |
| 14:02.12 | bjohnson | or you could filter based on callerid |
| 14:02.18 | bjohnson | authenticate them? |
| 14:02.29 | *** join/#asterisk isamar (~isamar@p8131-ipadfx21sasajima.aichi.ocn.ne.jp) |
| 14:02.34 | isamar | Hi folks... |
| 14:02.38 | MattH | bjohnson: username/password/auth user |
| 14:02.52 | bjohnson | not sure what you're talking about |
| 14:03.08 | MattH | bjohnson: you have to authenticate each phone you have.. or else it can't make calls... |
| 14:03.18 | isamar | I'm suffering a strange issue with chan_oh323 |
| 14:03.18 | dreamcode | is AbsoluteTimeOut aplies to every active channel idenpendtly or is it global ? |
| 14:03.32 | bjohnson | you mean the sip.conf config? |
| 14:03.57 | isamar | I use * with SIP phones and a remote h323 PSTN provider |
| 14:04.02 | MattH | bjohnson: yeah. |
| 14:04.04 | Gand_DJ | bjohnson, is there a way to call into the * box, and then somehow get a second dialtone to dial out. (such as dial in from pstn, and then dial out through * to a voip outbound line) |
| 14:04.24 | Gand_DJ | Guessing you can't setup that in amp |
| 14:04.26 | isamar | the SIP phones connect to me and I route their connection to a PSTN provider through H323.. |
| 14:04.34 | bjohnson | MattH: you config a phone in sip.conf and if you want to send calls to it you add them to extensions.conf. If you don't want to send calls to them, don't set them up in extensions.conf |
| 14:04.42 | isamar | all the phones has some config.. some dial plan/context... |
| 14:04.45 | MattH | Gand_DJ: no you can't.. you'd have to write a call plan for it |
| 14:04.59 | bjohnson | Gand_DJ: no idea for amp .. usual method to get a second dial tone is with disa |
| 14:05.06 | isamar | some are hanging up when the other side answers up the phone... |
| 14:05.10 | isamar | some not... |
| 14:05.25 | bjohnson | Gand_DJ: you should read the authenticate users page off the tips and tricks page on the wiki |
| 14:05.31 | *** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it) |
| 14:05.35 | Gand_DJ | k |
| 14:05.39 | bjohnson | Gand_DJ: you can actually use voicemail to do that too |
| 14:05.41 | MattH | bjohnson: ok makes sence |
| 14:05.41 | isamar | changing the PSTN provider to a SIP one.. it solves the problem... what it should be? |
| 14:05.54 | Gand_DJ | voicemail? |
| 14:06.00 | Gand_DJ | heh, how would that work |
| 14:06.15 | bjohnson | Gand_DJ: it's a setting in voicemail.conf |
| 14:06.57 | bjohnson | Gand_DJ: you dial in, go to voicemail, hit # I think, enter password, get access to voicemail menus, one is to go to another context which could let you dial out |
| 14:07.16 | *** join/#asterisk obihuan (~pepepep@80-28-2-2.adsl.nuria.telefonica-data.net) |
| 14:07.20 | obihuan | hello all |
| 14:07.21 | bjohnson | you could also authenticate via callerid |
| 14:07.28 | Gand_DJ | nextaftercmd=no? |
| 14:07.31 | Gand_DJ | change to yes? |
| 14:07.41 | Gand_DJ | learning all this as we go :) |
| 14:07.54 | isamar | hi obihuan... |
| 14:07.56 | Gand_DJ | Normally a system builder & computer consultant for locatl area. |
| 14:07.57 | Gand_DJ | lol |
| 14:07.57 | bjohnson | and there is an authenticate command that provides more flexibility than disa so most people use authenticate and then disa with no password |
| 14:08.22 | bjohnson | I don't remember |
| 14:08.39 | bjohnson | read the wiki page and edit it if you find you need to hunt for other info |
| 14:08.48 | *** join/#asterisk jmacz (~jmacz@63.245.86.140) |
| 14:09.22 | pjm_uk | gand_dj: btw I didnt find AMP suitable for pbx's with over abotu 30 or 40 extensions, due to the tedious method of adding extensions |
| 14:09.29 | pjm_uk | for small pbx's it is good though |
| 14:09.48 | bjohnson | it's good if it does exactly what you want it to do |
| 14:09.59 | obihuan | I have a PBX conected to a ISDN por on a Cisco Router to send all the extensions calls. I want to configure asterisk to replace the actual PBX, but i do not know how send the router the extensions calls. My dialplan it is ok. Any idea about what is going on? |
| 14:10.14 | Gand_DJ | I'm setting this up for a family voicemail system, and a second IVR for my SOHO. |
| 14:10.29 | pjm_uk | ok so for a few users AMP is a nice interface |
| 14:10.35 | MattH | pjm_uk: you could have written your own php "module" for amp to add them however you wanted.. though I find the backend database format of amp to be atroshious(sp)! |
| 14:10.44 | pjm_uk | but u have to add all users via the gui |
| 14:10.46 | Gand_DJ | Hoping to order a Sipura 3000 for pstn inbound, and also become a reseller for canada |
| 14:10.47 | bjohnson | unfortunately many people use it to avoid learning how to config * and then find something that they want to do that AMP doesn't handle. Then you're stuck with knowing nothing and trying to debug a VERY complex system .. good luck |
| 14:10.49 | km- | matth: atrocious |
| 14:10.54 | km- | matth: for your own personal edification |
| 14:10.54 | MattH | km-: yeah that's the word |
| 14:11.05 | MattH | :) |
| 14:11.08 | km- | :) |
| 14:11.10 | pjm_uk | ah yes matth i did think of parsing my existing extensions.conf into the amp dbase, but life is to short... |
| 14:11.20 | MattH | hehe |
| 14:11.27 | km- | matth: I'm not a spelling nazi, but if someone adds (sp) that usually elicits a response of the right spelling :) |
| 14:11.36 | MattH | hehe good enough for me |
| 14:11.50 | bjohnson | wright (sp) |
| 14:11.58 | *** join/#asterisk pif (ldm@zenon.apartia.fr) |
| 14:12.08 | MattH | lol |
| 14:12.12 | MattH | funny(sp) |
| 14:12.22 | MattH | :Þ |
| 14:12.24 | MeTaBSD | its possible to use the blacklist like --> database put blacklist 41855XXXXX 1 |
| 14:12.59 | bjohnson | no idea what youre talking about |
| 14:13.00 | km- | dammit |
| 14:13.09 | bjohnson | (sp) |
| 14:13.13 | km- | cant people stop calling so I can get this pending restart to work |
| 14:13.18 | bjohnson | damn it |
| 14:13.29 | km- | haha, that's spelling nazi! |
| 14:13.54 | km- | I could have typed it the "Right Way"[tm] but to what end? I'd type a bit more characters and you'd still get the same message across. :) |
| 14:14.04 | Essobi | SO CODE FOR JOO! NEXT! |
| 14:14.17 | km- | NO(sp) |
| 14:14.21 | km- | ;) |
| 14:14.36 | km- | hahhaahhahh |
| 14:14.51 | *** join/#asterisk JonR800 (jr@pcp05013027pcs.plyntv01.mi.comcast.net) |
| 14:14.57 | km- | I got the coolest piece of spam the other day |
| 14:15.03 | MattH | ? |
| 14:15.05 | km- | at first I thought it was cialis, etc |
| 14:15.17 | km- | but then I read it, and it was like a 15 page essay on why islam is the right religion |
| 14:15.30 | km- | I found it rather amusing |
| 14:15.35 | MattH | lol |
| 14:15.45 | Gand_DJ | When making IVR msg's for *, I'm guessing it would be best to use Ulaw to have the best quality prompts? |
| 14:15.51 | km- | only because nestled in between "enlarge your penis" and "CH34P V1AGR4" |
| 14:15.59 | Gand_DJ | I normally use gsm for softphone |
| 14:16.15 | km- | there's "INVITING CHRISTIANS TO THE RIGHT WAY AND REDEMPTION" |
| 14:16.17 | MattH | Gand_DJ: or record in a sound studie |
| 14:16.32 | km- | a room with carpeting is always god |
| 14:16.35 | km- | ... |
| 14:16.37 | km- | good |
| 14:17.02 | BrianR___ | or pay a few dollars and use thevoice... |
| 14:17.15 | ctooley | yes, don't record your prompts in the stall in the bathroom, they sound horrible |
| 14:17.21 | Gand_DJ | lol |
| 14:17.23 | MattH | lol |
| 14:17.42 | km- | haha |
| 14:17.58 | km- | the people at the office here think allison sounds too mechanical |
| 14:18.16 | BrianR___ | http://triggur.org/robodump/ <-- This guy's prompts were recorded in the bathroom stall.. |
| 14:19.21 | ctooley | use festival to record them all, then explain to them why Allison is better than festival |
| 14:19.31 | BrianR___ | FESTIVAL! |
| 14:19.49 | ctooley | Expectation Engineering should be a Masters program in college |
| 14:19.54 | BrianR___ | We have a hidden extension that plays "Thank you for calling the R J Reynolds tobacco company...." |
| 14:19.56 | ctooley | it's way more useful than MIS |
| 14:20.04 | BrianR___ | "Press 1 to hear more about my cancer kazoo" |
| 14:20.37 | BrianR___ | Yay! Sushi lunch finally... |
| 14:21.31 | ctooley | well, that certainly makes my Lean Pockets seem less exciting |
| 14:21.36 | bjohnson | km-: play them some of the funny ones |
| 14:21.59 | BrianR___ | 0 is still "ooooh! that tickles" on our asterisk test system... |
| 14:22.35 | bjohnson | Gand_DJ: I think most people just use gsm for the ivr prompts |
| 14:23.01 | Gand_DJ | ok |
| 14:23.30 | BrianR___ | We also have an "on hold forever" extension which has been real popular. |
| 14:23.33 | ctooley | bjohnson, don't you mean they persist the file in gsm? I think he was talking about to the phone that he's recording with. |
| 14:23.56 | MeTaBSD | its possible to use the blacklist like --> database put blacklist 41855XXXXX 1 |
| 14:23.58 | ctooley | Every time I do gsm to the phone the quality is choppy. |
| 14:27.18 | *** join/#asterisk ramtha (~tk@gw.01063telecom.de) |
| 14:28.29 | ManxPower | I feel hungover |
| 14:28.44 | ramtha | hi i have two TE410P in one box. how must the zaptel.conf looks like to activate the second card? |
| 14:29.26 | ManxPower | ramtha, the second card's channels start at 97 |
| 14:29.33 | ramtha | thx! |
| 14:30.06 | olivier_ | and 125 for E1 :) |
| 14:30.22 | ManxPower | ramtha, WHICH card is considered the "first card" is up to your motherboard |
| 14:30.36 | ManxPower | usually the card closest to the power supply will be considered the first card |
| 14:30.56 | km- | come on, one last call |
| 14:30.58 | km- | one last call... |
| 14:31.02 | km- | restart, restart, restart... |
| 14:31.29 | km- | yay! |
| 14:31.58 | julianjm | Hello, has anyone got an AVM Fritz! to work in ptp mode? If so, is mISDN needed? is it stable? what distro? |
| 14:34.27 | ramtha | hmm the last bchannel must be 235-249 by e1, right? |
| 14:35.09 | ramtha | ah 248 |
| 14:36.19 | newl | why 97? 4xE1 = 128 so shouldn't it be 129? :) |
| 14:37.41 | *** join/#asterisk af_ (~af@ip-148-227.sn1.eutelia.it) |
| 14:38.44 | obihuan | How could I connect asterisk to a router BRI port? and send calls to the router? I am not talkin about configure extensions.conf. I am talkin about especific config for the CAPI chanel.Please help needed |
| 14:39.20 | obihuan | now i get the following errors |
| 14:39.24 | obihuan | Connected to Asterisk 1.0.3 currently running on Asterisk-OKM (pid = 991) |
| 14:39.24 | obihuan | Verbosity is atleast 3 |
| 14:39.25 | obihuan | <PROTECTED> |
| 14:39.25 | obihuan | <PROTECTED> |
| 14:39.25 | obihuan | <PROTECTED> |
| 14:39.26 | obihuan | <PROTECTED> |
| 14:39.28 | obihuan | <PROTECTED> |
| 14:39.30 | obihuan | <PROTECTED> |
| 14:39.32 | obihuan | <PROTECTED> |
| 14:39.36 | obihuan | <PROTECTED> |
| 14:39.38 | obihuan | <PROTECTED> |
| 14:39.40 | obihuan | <PROTECTED> |
| 14:39.42 | obihuan | <PROTECTED> |
| 14:39.44 | obihuan | <PROTECTED> |
| 14:39.46 | obihuan | <PROTECTED> |
| 14:39.48 | newl | *smack* try pastebin.ca instead. ;) |
| 14:39.48 | obihuan | <PROTECTED> |
| 14:39.50 | obihuan | <PROTECTED> |
| 14:39.52 | obihuan | <PROTECTED> |
| 14:39.54 | obihuan | <PROTECTED> |
| 14:39.56 | obihuan | <PROTECTED> |
| 14:39.58 | obihuan | <PROTECTED> |
| 14:40.00 | obihuan | <PROTECTED> |
| 14:43.07 | obihuan | r |
| 14:43.20 | obihuan | any clue? |
| 14:43.29 | bjohnson | neither of us do |
| 14:44.52 | *** join/#asterisk grolloj (~chatzilla@slim-eth0.horizonlive.net) |
| 14:44.54 | *** join/#asterisk lbow (~steve@et0.cr2.ctn.nha.co.za) |
| 14:45.18 | Nugget | No one is available to supply clues at this time |
| 14:45.48 | yaboo | anyone owns the soyo n400s? |
| 14:45.59 | obihuan | I saw |
| 14:49.05 | [shodan] | yaboo, I was about to ask what was the catch with the n400s , but looks like no one here got one , probably because whoever got one no longer has any asterisk problem and isn't here asking for help ! |
| 14:50.41 | yaboo | shodan does it do sip or has fxo ports, or can you describe what features it has |
| 14:51.46 | [shodan] | the manual says there's a fxo version and a fxs version (who cares about the fxo version anyway , a pci fxo port is 6$ on ebay) , the manual also says it does sip |
| 14:53.04 | Gand_DJ | Hrm, with AMP, what is the difference between Digital Assistant and System recordings? (both prompt for voice recordings) |
| 14:53.07 | yaboo | shodan do you own one |
| 14:53.26 | *** join/#asterisk lbow (~steve@et0.cr2.ctn.nha.co.za) |
| 14:53.28 | [shodan] | nope |
| 14:53.29 | *** join/#asterisk fugitivo (~ajf@201.255.106.212) |
| 14:53.30 | fugitivo | hello |
| 14:53.37 | [shodan] | I'm sure there's a catch |
| 14:53.45 | zione | hello ! anyone using astguiclient and vicidial ? |
| 14:55.46 | yaboo | [shodan], hmm thinking off buying one |
| 14:56.19 | ManxPower | [shodan], You should care about FXO because 1) soon those cheap FXO ports will no longer be available and 2) you need each crd on it's own IRQ |
| 14:56.29 | bjohnson | hey .. got info on new Notel BCM50 "Built on Nortel Corporate Linux" |
| 14:57.07 | Hmmhesays | welltech is pissing me off |
| 14:57.30 | ManxPower | ~google site:lists.digium.com welltech crap |
| 14:57.41 | ManxPower | ~google site:lists.digium.com welltech problem |
| 14:57.53 | Hmmhesays | haha, i wasn't about to ask a question |
| 14:57.57 | Hmmhesays | I was just making a statement |
| 14:58.11 | eKo1 | What does welltech make? |
| 14:58.16 | Hmmhesays | junk |
| 14:58.23 | eKo1 | what kind of junk? |
| 14:58.28 | fugitivo | i'll need 2 fxo ports, any recommendation? |
| 14:58.32 | lbow | nufone users: are they working for you at the moment? I'm getting weird noises and my own voice echoed back... |
| 14:58.46 | Hmmhesays | I take that back, if you need something to set your beer on.... welltech is your answer |
| 14:58.51 | yaboo | ManxPower, what do i search on ebay for the x100p |
| 14:59.04 | ManxPower | yaboo, I don't know. |
| 14:59.04 | [shodan] | well , then the cheap x100p are gone , there will be 4 ports fxo n400s , but even then fxo ports are easier to make |
| 14:59.23 | [shodan] | yaboo, emm how about x100p |
| 14:59.30 | fugitivo | [shodan]: how much is the n400s? |
| 14:59.41 | [shodan] | 23.75$ |
| 14:59.43 | bjohnson | I wonder how much one of these bcm50 cost with all the options turned on |
| 14:59.43 | [shodan] | usd |
| 14:59.50 | yaboo | [shodan], so the n400s fxo units ain't out yet |
| 14:59.51 | fugitivo | [shodan]: with 4 ports? |
| 14:59.55 | [shodan] | yep |
| 15:00.00 | fugitivo | [shodan]: where? :) |
| 15:00.03 | [shodan] | so there's probably a catch |
| 15:00.11 | [shodan] | http://www.moselectronics.com/product/dhhubreaderrouter.html |
| 15:00.41 | bjohnson | fugitivo: 2 SPA 3000 |
| 15:01.06 | fugitivo | [shodan]: that's fxo or fxs?? |
| 15:01.22 | [shodan] | fxs |
| 15:01.35 | fugitivo | oh, you said fxo |
| 15:01.37 | fugitivo | i need fxo |
| 15:01.38 | fugitivo | 2 |
| 15:02.09 | [shodan] | ebay then |
| 15:02.16 | bjohnson | interesting .. has t.38 but no sip support |
| 15:02.19 | *** join/#asterisk bonez41 (~aint@c-67-166-77-14.hsd1.ut.comcast.net) |
| 15:02.22 | *** join/#asterisk bannerman (~bannerman@209.216.176.42) |
| 15:02.44 | fugitivo | but not the x100p, i need a more quality product for a small company |
| 15:03.01 | bjohnson | fugitivo: 2 SPA 3000 |
| 15:03.13 | bjohnson | or a digium with 2 fxo ports |
| 15:03.14 | Nuttah | ..x100p rocks |
| 15:03.40 | fugitivo | bjohnson: how much the spa? |
| 15:03.53 | bjohnson | $100 each at voxilla I think |
| 15:04.08 | bjohnson | one fxo plus one fxs |
| 15:04.09 | Hmmhesays | ok a normal sip invite message sent to a proxy should be sip:user@host or am I going nuts |
| 15:04.31 | bjohnson | [shodan]: that link you gave looks like a h323 device |
| 15:04.41 | fugitivo | bjohnson: and from digium what options do i have? |
| 15:05.31 | [shodan] | oh I tought I read sip somewhere ? what's the problem with h323 ? |
| 15:05.44 | Nuttah | its crap? |
| 15:05.53 | sambal | ~h323 |
| 15:05.54 | jbot | i guess h323 is An ITU-T standard for packet-based multimedia communications systems. This standard defines the different multimedia entities that make up a multimedia system - Endpoint, Gateway, Multipoint Conferencing Unit (MCU), and Gatekeeper - and their interaction. This standard is used for many voice-over-IP applications, and is heavily dependent on other ... |
| 15:05.57 | bjohnson | it's 10 year old tchnology |
| 15:06.16 | [shodan] | so it's well supported ? |
| 15:06.24 | Nuttah | heh.. not any more :P |
| 15:06.28 | Gand_DJ | bjohnson, It's funny that if you follow the eval units link from sipura, to the eval store hosted by voxilla, the SPA3k is $149 each |
| 15:06.41 | bjohnson | which in computer terms puts it on par to comparing a Athlon 2GHx with a 75 MHz Pentium |
| 15:06.44 | fugitivo | it's still used by voip providers in this country |
| 15:07.13 | [shodan] | well , it can do anything a pci fxs port can do , right ? I mean , it's not like it's going to talk to anything else than asterisk |
| 15:07.18 | Luhiwu | fugitivo? where are you from? Argentina? |
| 15:07.20 | Nuttah | its being weened out fugitivo |
| 15:07.28 | fugitivo | Luhiwu: yes |
| 15:07.38 | Luhiwu | fugitivo: drop IPlan :) |
| 15:07.58 | fugitivo | Luhiwu: heh, they're using h323 |
| 15:08.19 | Luhiwu | fugitivo: i know |
| 15:08.20 | bjohnson | [shodan]: h323 is a pita |
| 15:09.14 | Essobi | h323 is like T30.. |
| 15:09.16 | yaboo | bought 2 x 100p cards |
| 15:09.21 | BrianR___ | bjohnson: BCM 50, eh? |
| 15:09.26 | obihuan | How could I connect asterisk to a router BRI port? and send calls to the router? I am not talkin about configure extensions.conf. I am talkin about especific config for the CAPI chanel.Please help needed |
| 15:09.30 | BrianR___ | bjohnson: Does it still have gobs of gay licensing crap? |
| 15:09.35 | yaboo | was less than $50AUS |
| 15:09.40 | Essobi | The implementors can't get their assess out of their heads for long enough to spell the sunshine, and write a standard protocol stack. |
| 15:10.36 | Essobi | They all have their own version and flavor. |
| 15:10.43 | bjohnson | BrianR___: definitely .. that's why it's Norhell |
| 15:10.47 | [shodan] | bjohnson, well , if you compare setting up asterisk , compared to calling a "phone system consultant" or whatever , and just tell him "I want a kick ass phone system here's a blank check" , then asterisk is a pita to setup too , I mean , 23.75$ for 4xFXS or 305$USD for something similar http://store.yahoo.com/asteriskpbx/witd4pofxsbu.html |
| 15:11.12 | bjohnson | yaboo: see you later when you having trouble with them |
| 15:11.18 | [shodan] | is the TDM40B a 12-fold increase in convenience ? |
| 15:11.24 | Essobi | Tehe. |
| 15:11.33 | Nuttah | bjohnson: pff x100p i had no issues setting them up. |
| 15:11.36 | Essobi | Expect to have problems with 23.75 FXOs. |
| 15:11.49 | Essobi | Some people don't, but lots, do. |
| 15:12.02 | [shodan] | lots of people have trouble with windows xp |
| 15:12.02 | bjohnson | Nuttah: you're very, very lucky |
| 15:12.09 | Nuttah | i did get the cards from a digium reseller tho |
| 15:12.22 | [shodan] | lots have problem with programming vcrs |
| 15:12.26 | [shodan] | some facts would be nice |
| 15:12.34 | Essobi | [shodan] Lots of poeple are morons too. |
| 15:12.46 | Essobi | What's your point? |
| 15:12.49 | yaboo | bjohnson, trouble as in how |
| 15:13.06 | bjohnson | [shodan]: I guess my point is .. don't fool yourself into beliving it is something similar |
| 15:13.12 | Essobi | The knock off modems have about 5 different version of the same chipset, which only two work well with * but all the vendors sell them anyways. |
| 15:13.36 | Essobi | They don't give a shit.. what're you going to do? complain about a $12 modem? Like they care. |
| 15:14.03 | Essobi | and the knock off cards ARE physically different then the digium FXOs. |
| 15:14.05 | yaboo | Essobi, so I have a 40% chance of working |
| 15:14.17 | Nuttah | wildcard x100p are the ones i have |
| 15:14.29 | Essobi | yaboo Maybe.. I think the old chipset that worked well has been depreciated, another reason digium stopped carring the cards. |
| 15:14.39 | yaboo | Essobi, ok |
| 15:14.55 | [shodan] | well , first my fxos already work and I'm looking for 4 fxs and the guy who sell them actually supported them |
| 15:15.11 | yaboo | Essobi, this is what I bought |
| 15:15.15 | yaboo | http://cgi.ebay.com.au/ws/eBayISAPI.dll?ViewItem&category=70811&item=5768669493&rd=1&ssPageName=WDVW |
| 15:15.23 | bjohnson | yaboo: if you're looking for cheap hardware .. don't buy any. Not going to get cheaper than free |
| 15:15.23 | timecop | hmm |
| 15:15.24 | timecop | g723 is nice |
| 15:15.28 | timecop | but sounds like shit |
| 15:15.44 | Nuttah | aye.. use g711 ulaw almost entirely now |
| 15:15.59 | yaboo | bjohnson, true but who's handing them out here for free |
| 15:16.05 | Essobi | 729 is money. |
| 15:16.12 | [shodan] | I'm talking about soyo n400s 4x fxs vs TDM40B, in term of usability , what's the difference ? |
| 15:16.25 | bjohnson | yaboo: uhh .. damn near anyone will sell you nothing for free |
| 15:16.40 | bjohnson | yaboo: you could just use a voip provider |
| 15:16.51 | yaboo | bjohnson, in aus unlikely |
| 15:17.02 | timecop | OH |
| 15:17.05 | timecop | speaking of tdm400 |
| 15:17.12 | yaboo | will set up bjohnson my fxo on the sipura 3000 |
| 15:17.13 | bjohnson | hmm .. I saw a link to one offering free long distance a few months ago |
| 15:17.15 | timecop | do I *NEED* any modules on a tdm400 board to use zap features? |
| 15:17.17 | Nuttah | I like the SHIP from ASIA, with Norway written below it on that ebay item :P |
| 15:17.24 | timecop | or do I just need the board? |
| 15:17.34 | Essobi | timecop What what? |
| 15:17.51 | timecop | Essobi: i have a blank tdm400 board that I burned all the modules on |
| 15:17.53 | bjohnson | timecop: most people buy them with at least one module |
| 15:17.57 | timecop | duh |
| 15:18.02 | Essobi | timecop Youch. |
| 15:18.06 | timecop | but I just need it for meetme/zap timing |
| 15:18.09 | Essobi | so ... |
| 15:18.12 | timecop | so, does it work without any modules on it? |
| 15:18.13 | Essobi | Oh.. Don't do that. |
| 15:18.18 | bannerman | my iax debug is giving a lot of Tx-Frame-Retry and Rx-Frame Retry messages, and I'm occasionally losing outbound audio |
| 15:18.24 | Essobi | use the zapdummy |
| 15:18.41 | Essobi | on 2.6 kernel it's gravy.. |
| 15:18.56 | Essobi | 2.4X requires the right USB hardware to be present.. |
| 15:19.02 | bjohnson | [shodan]: go for it .. you're certain that we're all wrong. |
| 15:19.10 | timecop | i got intel usb, but last i heard it was kinda fucky with ztdummy |
| 15:19.17 | timecop | like not very accurate timing or somethign |
| 15:19.22 | yaboo | but Nuttah seen that before where there seller just has the factory in asia ship it |
| 15:19.24 | bjohnson | ztdummy I think |
| 15:19.41 | yaboo | worked that way when I bought the mvox 100 units |
| 15:19.41 | Essobi | timecop Use 2.6.. no USB needed then. |
| 15:20.03 | Nuttah | yaboo: i'm sure... but I have to agree with these guys. I dont recommend anything other than the wildcard x100p |
| 15:20.07 | timecop | who the fuck uses 2.6 on a production machine |
| 15:20.11 | kajtzu | timecop: a lot of people |
| 15:20.18 | timecop | nobody serious does |
| 15:20.18 | [shodan] | I think you have bias , not that you're wrong , you haven't said good about the tdm40b except that it's not the same thing |
| 15:20.22 | kajtzu | timecop: why not? |
| 15:20.22 | Essobi | timecop I DO!, I DO! |
| 15:20.25 | Essobi | :) |
| 15:20.29 | timecop | hell, 2.6 is slower on a bunch of machiens I had it on |
| 15:20.33 | timecop | disk access is way slower |
| 15:20.34 | kajtzu | timecop: RHEL4 works just fine |
| 15:20.44 | bjohnson | [shodan]: I also didn't recommend the tdm40b |
| 15:20.55 | Essobi | timecop Well.. Use FreeBSD.. Problem solved, AND you get safe disk writes. :) |
| 15:21.15 | timecop | i'll stick with 2.4 kthx |
| 15:21.16 | Nuttah | hmm which reminds me.. must try * on whitebox |
| 15:21.17 | Essobi | timecop I honestly havn't seen any slowdowns in the SCSI subsystem between 2.4 and 2.6. |
| 15:21.42 | [shodan] | then , hmm , I take that back then |
| 15:21.46 | Essobi | We.. are... talking about SCSI right? |
| 15:22.36 | Essobi | I mean... who uses IDE in a production environment? :) |
| 15:22.51 | Nuttah | still have a dtmf tone issue uusing my sipura units and DECT phones. |
| 15:22.54 | Essobi | Sure as hell ain't me. |
| 15:23.12 | *** join/#asterisk MichaelVanD (~MichaelVa@rrcs-24-123-121-190.central.biz.rr.com) |
| 15:23.25 | [shodan] | uh , I do |
| 15:23.33 | [shodan] | (use ide in production) |
| 15:23.50 | newl | love the speed and performance (hate the noise) but ya just can't knock the size of the IDE drives. |
| 15:23.50 | Nuttah | Essobi: nowadays a lot of people use IDE in production machines |
| 15:23.53 | newl | plus they're cheap as chips |
| 15:23.57 | Nuttah | well SATA really |
| 15:24.31 | *** join/#asterisk roamer323 (~sing@toronto-HSE-ppp4075335.sympatico.ca) |
| 15:25.02 | [shodan] | with ide you get low cost per gb , that = redundancy , redundancy = speed and reliability |
| 15:25.08 | *** join/#asterisk makbut (~christian@200.121.129.178) |
| 15:25.29 | newl | erm..no |
| 15:25.52 | *** join/#asterisk heison (~heison@gw-yyz.somanetworks.com) |
| 15:27.17 | Gand_DJ | Seagate SATA HD's with NCQ provide almost as much performace as 10k raptor drives, |
| 15:27.43 | Gand_DJ | When optimized within a Raid environment |
| 15:28.18 | Gand_DJ | Command queueing for the drives is only upto 32 commands, whereas SCSI is upto 256 commands |
| 15:28.59 | [shodan] | I build a pair of 12 drive (seagate 200gb) server , 4.8tb in raid5 , 30% loss of space but you can have up to 4 drives failling, and when you're reading it's taking data from multiple drives at once increasing speed , how much does it cost to have scsi raid5 of 3.3tb ? my setup cost 5500$cad |
| 15:29.13 | DrWho17 | sorry I've got several 3ware IDE arrays, been working fine for years |
| 15:29.19 | TheEmperor | when i remotely ssh into my * box, how to i exit (i'm using putty) and still leave * running? |
| 15:29.28 | DrWho17 | raid10 is nice |
| 15:29.31 | *** join/#asterisk UBiQUiTY (~mike@68.160.103.76) |
| 15:29.41 | newl | sure, but what is the true (read production environment) MTBF for IDE when compared to SCSI? |
| 15:29.43 | Gand_DJ | Yeah, but raid 10 = 50% loss of HD |
| 15:29.58 | Makenshi | scsi is slow... fc disks better :> |
| 15:29.59 | Gand_DJ | Huge failsafe though |
| 15:30.22 | DrWho17 | Gand_DJ: yea, I don't care about that, I can buy many IDE drives for the equivalent SCSI storage |
| 15:30.30 | newl | TheEmperor: type exit. |
| 15:30.35 | Makenshi | and the least i would use is sata, never ide |
| 15:30.38 | DrWho17 | I have 2 hot spares on each of those RAID 10's as well |
| 15:30.49 | TheEmperor | newl:Ii did that, and then it goes to the command line interface |
| 15:30.50 | Makenshi | i have a 2tb array using sata disks |
| 15:30.50 | [shodan] | mtbf isn't important with 12 drives (well unles you compare it to 12 drive scsi but since low cost=redundancy , mtbf is just part of the cost) |
| 15:30.54 | Makenshi | 2 hot spares |
| 15:31.13 | Makenshi | leaving 6 usable disks, one of which is parity for a raid5 array |
| 15:31.15 | newl | TheEmperor: type it again, one gets you out of the asterisk CLI, the second will get you out of the shell. |
| 15:31.17 | Gand_DJ | [shodan], I sell Seagate 200GB SATA/NCQ 7200RPM HD's for$175CDN each |
| 15:31.25 | Makenshi | my disks are 250gb |
| 15:31.29 | DrWho17 | Makenshi: good for you, most SATA drives are just IDE drives with an adapter chip, mechanically the drives are the same |
| 15:31.35 | [shodan] | Gand_DJ I buy then for 132$cad |
| 15:31.40 | TheEmperor | newl: still always goes back to CLI... |
| 15:31.49 | [shodan] | (eprom.com) |
| 15:31.49 | Makenshi | DrWho17, yes, but at least the performance isn't as shoddy :) |
| 15:31.52 | TheEmperor | newl: then I get this |
| 15:31.54 | DrWho17 | MTBF is the same, and performance is the same |
| 15:31.58 | TheEmperor | newl:The QUIT and EXIT commands may no longer be used to shutdown the PBX. |
| 15:31.58 | TheEmperor | Please use STOP NOW instead, if you wish to shutdown the PBX. |
| 15:32.02 | [shodan] | DrWho17, not seagates |
| 15:32.05 | DrWho17 | Makenshi: only with the very latest chipsets |
| 15:32.17 | *** join/#asterisk heison (~heison@ns.somanetworks.com) |
| 15:32.21 | DrWho17 | and pc's |
| 15:32.36 | eKo1 | TheEmperor: You need to start * as a daemon process. |
| 15:32.48 | TheEmperor | eko1:how do i do that? |
| 15:32.53 | newl | heh yeah, that was going to be the next question. :) |
| 15:32.53 | eKo1 | just type asterisk |
| 15:33.07 | *** join/#asterisk DeeJayTwo (~deejay2@office.abi.ca) |
| 15:33.08 | DeeJayTwo | hi |
| 15:33.15 | TheEmperor | ? |
| 15:33.18 | TheEmperor | that's it? |
| 15:33.20 | Gand_DJ | [shodan], if you are a dealer, then yeah.. because that is dealer price |
| 15:33.21 | eKo1 | yep |
| 15:33.22 | Nuttah | aye |
| 15:33.26 | DeeJayTwo | I'm trying to connect a TA-750 channel bank to a digium TE410P (quad t1 port)... |
| 15:33.28 | Nuttah | asterisk -r to reconnect to the cli |
| 15:33.34 | DeeJayTwo | ztcfg -vvv shows everything is ok for channels.. |
| 15:33.44 | DeeJayTwo | but the lights on the card are still glowing red.. |
| 15:33.51 | [shodan] | Gand_DJ, yeah I know , but they sell to any company , not just dealers , they're great ! |
| 15:33.54 | DeeJayTwo | is the T1 wire the same as a straight network cable? |
| 15:34.16 | DeeJayTwo | it's a 6 foot cat-5e RJ-45 cable.. |
| 15:34.45 | tzanger | DeeJayTwo: yes if it's straight through |
| 15:35.22 | tzanger | DeeJayTwo: if yuou need a T1 cross then no a cat5 cross will NOT work |
| 15:35.22 | Gand_DJ | [shodan], who do you use for voip? I'm in manitoba.. still deciding who to use for * outbound & inbound |
| 15:35.37 | TheEmperor | eko1:doesn't seem to work.. |
| 15:35.49 | TheEmperor | eko1:i logged in as root into my server, typed in asterisk |
| 15:35.57 | [shodan] | Gand_DJ, there's no voip provider in my area, so I use the pots |
| 15:36.08 | TheEmperor | eko1: and the asterisk -r, but when i shutdown the panel, asterisk is shutdown as well |
| 15:36.31 | Gand_DJ | [shodan], you use * just for internal extensions, ivr, and voicemail? |
| 15:37.21 | [shodan] | yep , well it's not done implementing , I haven't got fxs yet , but voice mail, pager and all that pbx stuff is great by itself |
| 15:37.28 | Nuttah | TheEmperor: how did you disconnect from the cli after astrisk -r? |
| 15:37.32 | [shodan] | anyway my WISP is too shoddy for voip |
| 15:37.38 | TheEmperor | Nuttah:exit |
| 15:37.54 | Gand_DJ | [shodan] lol.. using a WISP here also... since I'm in the country |
| 15:37.57 | Nuttah | then I dont know.. that should only disconnect you.. not stop asterisk |
| 15:38.04 | Gand_DJ | 2mbps upload/download.. 3gb month limit |
| 15:38.20 | Gand_DJ | pretty crappy |
| 15:38.30 | [shodan] | same thing here , (well 6gb) and they charge 5$ per extra gb ! |
| 15:38.34 | eKo1 | I work for a WISP and 2Mbps is unheard of. |
| 15:38.49 | Gand_DJ | heh.. $10 / gb here.. unless I prebook extra gb... then it's $7 / gb |
| 15:38.59 | [shodan] | eKo1, you mean , too low or too high ? |
| 15:39.05 | eKo1 | too high |
| 15:39.36 | [shodan] | eKo1, my connection is actually working a 4mbps , it's canopy crap , but they screwed up the flow control mahahaha ! |
| 15:39.36 | eKo1 | of course, the ISP only has a 4 Mbps link to the internet so... |
| 15:39.49 | Gand_DJ | My WISP is using Waverider modems |
| 15:39.56 | Gand_DJ | They max at 2mbps each way |
| 15:40.30 | [shodan] | canopy can do 10mpbs but their network is improperly bridged and there is a -lot- of noise |
| 15:41.06 | tzanger | bkw_: I have plenty to bitch about, I just can't make it to the dev conf :-) |
| 15:41.14 | Gand_DJ | They are using higher model modems in the city, which can do upto 70mbps+, for the right price |
| 15:41.23 | [shodan] | also their bandwidth counter crash when a pppoe connection goes over 700mb :)) |
| 15:41.35 | eKo1 | We have a customer right now who is bitching about lack of bandwidth. |
| 15:41.48 | [shodan] | when the connection closes the bandwidth it doesn't get counted hehe |
| 15:41.55 | [shodan] | eKo1, how much do you charge ? |
| 15:42.13 | [shodan] | flat fee ? or per gb ? |
| 15:43.13 | eKo1 | $320 for a 256 kbps flat fee. |
| 15:43.23 | *** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230) |
| 15:43.29 | [shodan] | oh my god ! |
| 15:43.31 | eKo1 | Of course, the customer never really gets 256 kbps so... |
| 15:43.35 | heison | ~seen moc[toronto] |
| 15:43.39 | jbot | moc[toronto] is currently on #asterisk (16h 48m 58s). Has said a total of 7 messages. Is idling for 10h 41m 7s |
| 15:43.44 | [shodan] | this is the worst I ever heard ! ;) |
| 15:43.46 | newl | I'll keep my 8Mbit ADSL for $90AUD thanks. :) |
| 15:44.01 | [shodan] | and -I- thought I had it bad ! |
| 15:44.04 | eKo1 | Bandwidth here is mad expensive. |
| 15:44.29 | [shodan] | I mean , this is pass-your-own-fiber-instead expensive ! |
| 15:44.30 | eKo1 | Oh, and the installation costs $2000 |
| 15:44.38 | R3DB0x | eKol what kind of wireless gear do u use....cause we offer 3Mbit on our wireless system |
| 15:44.42 | [shodan] | satellite is way cheaper than that ! |
| 15:44.57 | eKo1 | We use EZ bridge radios. |
| 15:45.07 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com) |
| 15:45.08 | newl | anywho..sleepies. 'night |
| 15:45.16 | R3DB0x | why so high on the install |
| 15:45.20 | [shodan] | eKo1, what range ? 150km ? |
| 15:45.38 | eKo1 | yeah right, try 30 |
| 15:45.47 | eKo1 | at most |
| 15:45.57 | [shodan] | eh , ezbridge stinks ! |
| 15:46.09 | eKo1 | R3DB0x: beats me. I don't come up with prices. |
| 15:46.17 | Hmmhesays | anyone else have equipment that sends <sip:{username}@{uac_ip}> ? |
| 15:46.40 | Hmmhesays | heh, forgot some info in that sentence |
| 15:46.58 | Hmmhesays | in the sip invite message I have one piece of equipment that is senting <sip:{username}@{uac_ip}> in the from field |
| 15:47.02 | DeeJayTwo | is there a howto somewhere on how to setup a channel bank with asterisk? |
| 15:47.15 | tzanger | DeeJayTwo: there isn't much to it |
| 15:47.17 | Hmmhesays | instead of <sip:{username}@{server_ip}> |
| 15:47.17 | DeeJayTwo | with fxo at the other side of the channel bank (serving clients with phones) |
| 15:47.28 | *** join/#asterisk nrc (~username@zeus.eurotux.com) |
| 15:47.30 | DeeJayTwo | with a straight cable, no link yet... |
| 15:47.39 | eKo1 | Try another cable. |
| 15:47.42 | DeeJayTwo | I tested it.. |
| 15:47.50 | DeeJayTwo | parallel, 8 pins up |
| 15:48.09 | eKo1 | You aren't listening. Try another cable. |
| 15:48.11 | tzanger | DeeJayTwo: configure the TE110P as the clock source (sync=0) and make the span b8zs/esf |
| 15:48.40 | Hmmhesays | so I'm kind of curious if this is an invalid way to send a sip invite |
| 15:48.50 | *** join/#asterisk loutux (oooooooo@200.124.234.228) |
| 15:48.55 | *** join/#asterisk Ahewes (~rsb@adsl-69-107-53-145.dsl.pltn13.pacbell.net) |
| 15:48.56 | tzanger | DeeJayTwo: then just set up zapata to use the right signaling (fxs or fxo) and that should be all there is to it |
| 15:49.03 | loutux | hi everybody! |
| 15:49.11 | eKo1 | Hmmhesays: as long as the devices see each, doesn't matter. |
| 15:49.20 | DeeJayTwo | tzanger..it's already that way.. |
| 15:49.27 | DeeJayTwo | span=1,0,0,esf,b8zs |
| 15:49.27 | DeeJayTwo | fxoks=1-24 |
| 15:49.34 | DeeJayTwo | that's what is in my zaptel.conf |
| 15:49.51 | Ahewes | For anyone using a Sipura 841, I've got a few questions. |
| 15:49.56 | DeeJayTwo | should I use something else than "ks" ? |
| 15:50.06 | Hmmhesays | well, this guy with a welltech is telling me that <sip:{username}@{uac_ip}> in the from field is invalid, that's why his proxy is rejecting the invite |
| 15:50.28 | eKo1 | What IP is it expecting? |
| 15:50.42 | Ahewes | Is it possible to do that "on-hook paging" think with a sipura 841, where you call the 841, send a special sip message, and the speakerphone comes off hook? |
| 15:50.56 | Hmmhesays | well, this guy with a welltech is telling me that <sip:{username}@{proxy ip}> in the from field is invalid, that's why his proxy is rejecting the invite |
| 15:50.59 | Hmmhesays | oops |
| 15:51.01 | tzanger | DeeJayTwo: ok so it's an FXS channel bank? |
| 15:51.16 | Hmmhesays | He says he is expecting <sip:{username}@{proxy ip}> |
| 15:51.17 | DeeJayTwo | yes |
| 15:51.24 | tzanger | DeeJayTwo: ok, so what seems to be the trouble? |
| 15:51.43 | DeeJayTwo | the T1 link doesn't gets up.. |
| 15:51.47 | DeeJayTwo | ;) |
| 15:51.54 | tzanger | DeeJayTwo: plug a loopback cable in |
| 15:51.56 | tzanger | does it go green? |
| 15:52.06 | tzanger | are you using a t1 crossover cable? |
| 15:52.17 | DeeJayTwo | it's a straight (parallel) rj-45 cable |
| 15:52.19 | DeeJayTwo | tested.. |
| 15:52.20 | DeeJayTwo | 8 pins.. |
| 15:52.21 | tzanger | DeeJayTwo: won't work |
| 15:52.25 | tzanger | DeeJayTwo: you need a T1 crossover |
| 15:52.31 | Hmmhesays | So i'd like to tell him he's full of crap, but I can't find the info in the rfc |
| 15:52.32 | tzanger | DeeJayTwo: NOT an ethernet crossover either |
| 15:52.34 | DeeJayTwo | to connect TE410P to Channel bank? |
| 15:52.37 | tzanger | DeeJayTwo: yup |
| 15:52.43 | DeeJayTwo | duh.. |
| 15:52.51 | Hmmhesays | well.. not "can't find it" ... I just haven't found it yet |
| 15:52.51 | DeeJayTwo | where could I find such cross over schema? |
| 15:52.52 | DeeJayTwo | to make it.. |
| 15:52.52 | tzanger | pin 1 -> 4, pin 2 -> 5 |
| 15:52.56 | DeeJayTwo | ok |
| 15:53.03 | tzanger | DeeJayTwo: basically pairs 1&2 are crossed |
| 15:53.13 | tzanger | ethernet uses pairs 2&3 |
| 15:53.17 | eKo1 | google for it |
| 15:53.32 | tzanger | which is why an ethernet cross won't work, but an ethernet patch cable owrks as a T1 patch :-) |
| 15:53.42 | DeeJayTwo | ok nice =) |
| 15:53.49 | PatrickDK | tzanger, no |
| 15:53.59 | tzanger | PatrickDK: what |
| 15:54.03 | Moonwick | ethernet uses 1, 2, 3, and 6 |
| 15:54.03 | PatrickDK | it won't work for t1, t1 needs to be shilded |
| 15:54.12 | tzanger | PatrickDK: for DSX1 it works just fine |
| 15:54.15 | Moonwick | swap 1->3 and 2->6 |
| 15:54.22 | tzanger | Moonwick: that's what I said |
| 15:54.22 | AgiNamu | anyone here have success with getting asterisk to create a gmon.out file? |
| 15:54.25 | AgiNamu | I compiled with -pg |
| 15:54.28 | PatrickDK | and it depends what ethernet standard your using |
| 15:54.32 | PatrickDK | 586b or 586a |
| 15:54.41 | tzanger | PatrickDK: nobody uses anything other than cat5 or 5e these days |
| 15:54.51 | Moonwick | no, because pin 5 isn't used in ethernet :) |
| 15:55.00 | tzanger | PatrickDK: pair 1 = pins 4&5, pair 2 = pins 1&2, pair 3 = pins 3&6, pair 4 = pins 7&8 |
| 15:55.19 | PatrickDK | tzanger, only on 586b |
| 15:55.21 | Moonwick | ah |
| 15:55.27 | tzanger | Moonwick: I said ethernet was pairs 2&3, and T1 was pairs 172 |
| 15:55.29 | tzanger | er 1&2 |
| 15:55.34 | Gand_DJ | How do you guys get MusicOnHold to stream music from the internet instead of using mp3s? |
| 15:55.38 | Moonwick | ah, ignore me then :) |
| 15:55.39 | tzanger | PatrickDK: PatrickDK well 568a is fucked up anyway |
| 15:56.08 | tzanger | PatrickDK: I think 568a was created by the data crowd just to feel special :-) |
| 15:56.21 | PatrickDK | na, 586a is how they used to do it, based on usco |
| 15:56.35 | PatrickDK | but they figured out 586b pairs have less interferance |
| 15:56.37 | PatrickDK | than 586a pairs |
| 15:56.40 | PatrickDK | so they switched |
| 15:56.46 | ChkDigit | GrandDJ - Static content, or live? |
| 15:56.50 | tzanger | I just go based on telco for both data and voice... in which pair 1 is middle, pair 2 is end, 3 straddles middle and 4 is opposite end |
| 15:57.05 | Gand_DJ | live..... as static stuff I guess you can just save as mp3 format |
| 15:57.10 | *** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it) |
| 15:57.13 | Gand_DJ | Like for Digitally Imported or something |
| 15:57.15 | bjohnson | this nortel bcm50 might be okay to tie nortel handsets into a voip system .. looks like each bcm can handle up to 12 of the nortel digital handsets |
| 15:57.26 | *** join/#asterisk wiz8291 (~dang@kay.arcbox.com) |
| 15:57.28 | wiz8291 | hey guys |
| 15:57.34 | wiz8291 | how do i place a call on hold? |
| 15:57.45 | wiz8291 | is parking the only way? |
| 15:57.58 | PatrickDK | musiconhold() |
| 15:58.16 | wiz8291 | i mean from the phone |
| 15:58.17 | tzanger | PatrickDK: ... so they went from 568b and created 568a? seems backward |
| 15:58.21 | bjohnson | Gand_DJ: live internet streams are 1. not very reliable and 2. add up if you have bandwidth caps |
| 15:58.27 | wiz8291 | i.e. *2 |
| 15:58.34 | PatrickDK | no, from 586, to 586b, and renamed 586 to 586a |
| 15:58.38 | bjohnson | Gand_DJ: that being said, I find that mplayer handles streams better than mpg123 |
| 15:58.49 | ChkDigit | GrandDJ - I'm just guessing, as I haven't done it, but you'll have to have mpg123 or the like playing music continuously to a program that will send audio out to the MusicOnHold process whenever it doesn't block. |
| 15:58.51 | tzanger | PatrickDK: ahh |
| 15:58.53 | eKo1 | well, mpg123 is known to suck |
| 15:59.00 | tzanger | PatrickDK: good history lesson; thank you :-) |
| 15:59.12 | *** join/#asterisk Meaty (~patate@office.abi.ca) |
| 15:59.19 | PatrickDK | I still like shielded wire, it's good stuff |
| 15:59.21 | Gand_DJ | I have the mpg123 program that comes with *@home :) |
| 15:59.24 | tzanger | I'm curiosu though as to how 568b has less interferance than a... they didn't adjust pairing, just which was used first, no? |
| 15:59.37 | bjohnson | Gand_DJ: we use slimserver for a web interface to select local mp3 files or switch to radio streams .. then use mplayer to play the stream from the local server |
| 15:59.42 | PatrickDK | tzanger, if every pair was twisted the same, it wouldn't matter if you twisted them |
| 15:59.48 | tzanger | PatrickDK: they're not though |
| 15:59.48 | PatrickDK | each pair is twisted a different amount |
| 16:00.02 | ChkDigit | GrandDJ - the problem is that when nobody is listening to music on hold, asterisk pauses it (uhhh, blocks the input stream...) |
| 16:00.08 | PatrickDK | I believe they are using the higher twist count pairs |
| 16:00.14 | bjohnson | Gand_DJ: I found I had to kill and restart mpg123 hourly when playing streams .. mplayer only once a day |
| 16:00.17 | PatrickDK | I would have to double check though |
| 16:00.20 | tzanger | PatrickDK: right, otherwise they would cross-couple. they're twisted to keep the induced noise the same on the pairs, but each pair is twisted slightly differently to keep pair-pair crosstalk down |
| 16:00.34 | wiz8291 | anyone? |
| 16:00.51 | PatrickDK | I know the blue pair isn't twisted much at all, compaired to the others |
| 16:00.58 | tzanger | so they discovered that pair1 and 2's twist in 568a was a little better than 1&3 |
| 16:00.59 | PatrickDK | I used to know exactly how many twists there are |
| 16:01.00 | Gand_DJ | hrm, so mplayer will work for *, and it connects to an m3u file? |
| 16:01.06 | tzanger | blue is pair 1 |
| 16:01.13 | bannerman | Is the "testyourvoip.com" test worth anything? |
| 16:01.20 | tzanger | er no |
| 16:01.22 | tzanger | no yes |
| 16:01.23 | tzanger | it si |
| 16:01.30 | PatrickDK | blue is always pair1 |
| 16:01.33 | tzanger | blue orange green brown I think... it's been a while |
| 16:01.35 | PatrickDK | 586b uses 2 and 3 |
| 16:01.39 | eKo1 | I already know my voip sucks so... |
| 16:01.48 | ChkDigit | Gand_DJ - Yup, but if noone uses music on hold, it may timeout the live connection. |
| 16:01.56 | Gand_DJ | ok. |
| 16:01.58 | bjohnson | Gand_DJ: well .. I actully don't feed mplayer into * but out the headphone jack and into our Nortel. But should work with * I think |
| 16:02.11 | tzanger | blue orange green brown slate I think and hten the binders are like white yellow red black violet |
| 16:02.14 | tzanger | I think... |
| 16:02.14 | tzanger | heh |
| 16:02.44 | Gand_DJ | ok. i'll probably stick to standard mp3's |
| 16:04.23 | Gand_DJ | Does mpg123 randomly play the files, or does it play them in sequence starting with mp3 #1 each time sub is on hold |
| 16:06.14 | PatrickDK | seqence, by default |
| 16:06.44 | Gand_DJ | heh.. so everytime i put a guy on hold he'll hear the same song over & over...lol |
| 16:06.52 | PatrickDK | no |
| 16:07.02 | PatrickDK | it starts from where the last guy was taken off hold |
| 16:07.16 | Gand_DJ | ok. that's good |
| 16:07.56 | jbragnar | i have a problem with sound quality, the sound in the phones connected to asterisk is "bubbly" after a while |
| 16:08.05 | jbragnar | the other side is fine |
| 16:08.26 | jbragnar | how do i debug this? are there any docs? |
| 16:09.04 | bjohnson | I have a problem with sound quality, when I drink too much "bubbly" I sound funny |
| 16:09.23 | jbragnar | bjohnson: just get off the phone then |
| 16:09.42 | jbragnar | shouldnt drink and phone at the same time anyway |
| 16:09.54 | bjohnson | oh .. I wondered what the hard thing was I was sitting on |
| 16:10.20 | DeeJayTwo | I have some parameters I don't know what to do with in the channel bank.. |
| 16:10.35 | Meaty | Hi DeeJay Two |
| 16:10.44 | DeeJayTwo | T1 Timing / T1 CSU Loopback / T1 SCL-96 digigroup search |
| 16:10.53 | DeeJayTwo | and T1 Count |
| 16:11.09 | *** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) |
| 16:11.11 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-204-197.dsl.scarlet.be) |
| 16:11.12 | DeeJayTwo | Framing and line code is: esf, b8zs, which I think is correc.t. |
| 16:13.36 | IPmonger | those are pretty standard |
| 16:14.33 | DeeJayTwo | what about timing ? Loop/Local/External |
| 16:14.34 | DeeJayTwo | ?! |
| 16:14.39 | IPmonger | loop |
| 16:14.44 | IPmonger | that means you get it from the telco |
| 16:14.47 | *** join/#asterisk HiroProtagonist (~hiro@64.78.237.253) |
| 16:15.01 | IPmonger | local means you use your internal [crappy] clock |
| 16:15.04 | DeeJayTwo | this is a fxs channel bank.. |
| 16:15.13 | IPmonger | external means you bought and expensive timing source |
| 16:15.18 | DeeJayTwo | ok |
| 16:15.31 | DeeJayTwo | so if I set it to loop... the pbx must provide the timing? |
| 16:15.45 | DeeJayTwo | T1 count: D4 count.. |
| 16:15.46 | DeeJayTwo | is it ok? |
| 16:15.53 | DeeJayTwo | there's D1D also.. |
| 16:16.10 | HiroProtagonist | I am trying to set up asterisk to accept incoming VoIP calls and then forward them to my cellphone using POTS...is there any documentation out there that explains how to easily accomplish this? |
| 16:16.59 | *** join/#asterisk stevej (~stevej@67.97.36.243) |
| 16:16.59 | IPmonger | DeeJayTwo: yes, loop means coming in off the line [from whatever is at the other end] |
| 16:17.01 | Gand_DJ | HiroProtagonist, What do you use for pstn? |
| 16:17.06 | DeeJayTwo | ok..so in asterisk |
| 16:17.09 | DeeJayTwo | how do I setup the timing? |
| 16:17.13 | Gand_DJ | I'm looking at an sipura 3000 possibly |
| 16:17.14 | AgiNamu | HiroProtagonist, have you read the sample config files? |
| 16:17.21 | Moonwick | well, you might try googling for "sending-hiroprotagonist's-voip-calls-to-his-cellphone-HOWTO" |
| 16:17.31 | Moonwick | failing that, voip-info.org is a good resource |
| 16:17.34 | HiroProtagonist | Gand_DJ, I'm still open, preferably something cheap, like the IA92 |
| 16:17.36 | tzanger | DeeJayTwo: nine times out of ten the PBX will not NOT sync to the line; they're generally designed to do that since generally they hook up to a telco switch. |
| 16:17.47 | HiroProtagonist | AgiNamu, no, I haven't yet |
| 16:17.55 | AgiNamu | do that first. |
| 16:17.57 | Gand_DJ | HiroProtagonist, you'd need an fxo device for making calls out to pstn |
| 16:17.59 | AgiNamu | it'll teach you a lot |
| 16:18.02 | DeeJayTwo | ok.. |
| 16:18.06 | tzanger | DeeJayTwo: I tend to tell the channel banks to not sync since they ahve a better (more stable) clock than I believe the cheap TJ320 chip oscillator can provide |
| 16:18.14 | tzanger | but ofr now let asterisk provide sync |
| 16:18.23 | DeeJayTwo | ok.. |
| 16:18.33 | HiroProtagonist | Gand_DJ, would the IA92 softmodem work well for this? |
| 16:18.36 | DeeJayTwo | is span=1,0,0,esf,b8zs still correct? |
| 16:18.47 | DeeJayTwo | or do I need to change the second parameter? |
| 16:18.48 | tzanger | if they (asterisk and the channel bank) conflict in this setting it WILL generally work, but you will get the occassional 'buzz' or 'chirp' as you get a frame slip. |
| 16:18.55 | DeeJayTwo | ok.. |
| 16:19.01 | JerJer | second parameter |
| 16:19.03 | Gand_DJ | HiroProtagonist, possibly... but alot of people are saying the sound quality & echo is bad |
| 16:19.07 | tzanger | DeeJayTwo: that line is telling asterisk NOT to sync ot the remote side. That is fine, set your CB to sync ot the span |
| 16:19.22 | Gand_DJ | seems people like the sipura 3000 as it has proper echo cancellation and good voice quality |
| 16:19.33 | DeeJayTwo | tzanger: External ? |
| 16:19.37 | DeeJayTwo | or loop? |
| 16:19.40 | DeeJayTwo | (to achieve it) |
| 16:19.40 | tzanger | DeeJayTwo: loop |
| 16:19.44 | DeeJayTwo | ok perfect.. |
| 16:19.46 | DeeJayTwo | it's already loop.. |
| 16:19.50 | DeeJayTwo | so made the wire.. |
| 16:19.52 | tzanger | external implies a third device plugged in somewhere (stratum 4 or better clock source) |
| 16:19.54 | HiroProtagonist | Gand_DJ, what would you recommend, and how much would that cost me? |
| 16:19.56 | DeeJayTwo | with 1->4 2->5.. |
| 16:19.58 | DeeJayTwo | it doesn't work...:( |
| 16:20.04 | Gand_DJ | well I don't have any fxo device myself |
| 16:20.05 | tzanger | DeeJayTwo: the TE110P is coming up red? |
| 16:20.11 | DeeJayTwo | TE410P |
| 16:20.15 | tzanger | well whatever |
| 16:20.16 | DeeJayTwo | it's glowing red.. |
| 16:20.17 | tzanger | it's coming up red? |
| 16:20.20 | HiroProtagonist | Is anybody using LineJack? |
| 16:20.22 | Gand_DJ | You can always get a soft modem to test it. |
| 16:20.28 | DeeJayTwo | the loopback wire make it turn green |
| 16:20.29 | JerJer | HiroProtagonist: no and you shouldn't either |
| 16:20.36 | tzanger | DeeJayTwo: ok. Make a loopback PLUG. just an RJ11 with 2 wires in it. one going from 1 to 4 and one going from 2 to 5 |
| 16:20.46 | HiroProtagonist | Okay, so what should I use for fxo? |
| 16:21.11 | tzanger | pin 1 is on the left hand side when the RJ45 jack is pointing upward (with the pins at the top facing you and the cable (if it were there) coming "up" into the bottom of the connector |
| 16:21.47 | tzanger | so take a pair of wires and stick them in to 1&2, then bend them back so that it's liek this |
| 16:21.57 | tzanger | 12.12... |
| 16:22.06 | tzanger | where 1 and 2 are your wires #1 and #2 :-) |
| 16:22.12 | *** join/#asterisk Himeko (~himeko@S01060040ca128fc3.ed.shawcable.net) |
| 16:22.14 | tzanger | or if you're using standard colours |
| 16:22.29 | tzanger | bluewhite, blue, blank, bluewhite, blue, blank, blank, blank |
| 16:22.49 | tzanger | again that order is with the connector facing "up" and the wires looping out the bottom |
| 16:23.01 | DeeJayTwo | good it works.. |
| 16:23.07 | DeeJayTwo | I have 2 channel banks.. |
| 16:23.09 | IPmonger | yay! |
| 16:23.11 | DeeJayTwo | one doesn't work.. (it seems) |
| 16:23.18 | DeeJayTwo | I will try to see what's wrong in the conf.. |
| 16:23.23 | DeeJayTwo | on the other one.. |
| 16:23.37 | tzanger | DeeJayTwo: are they both the same CB? (Which CB?) |
| 16:24.44 | Gand_DJ | ne1 successfully setup * to handle inbound and outbound for firefly network? |
| 16:24.46 | Gand_DJ | freshtel |
| 16:25.16 | zoa | http://www.asteriskguru.com/tools/bandwidth/index.php :) |
| 16:25.21 | *** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it) |
| 16:29.06 | *** join/#asterisk EvlHimeko (~himeko@S01060040ca128fc3.ed.shawcable.net) |
| 16:45.36 | *** join/#asterisk lattice (~lattice@S010600045ad57bb6.vc.shawcable.net) |
| 16:47.26 | *** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net) |
| 16:47.27 | *** join/#asterisk juice (~juice@mo-65-173-76-11.dyn.sprint-hsd.net) |
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| 16:49.38 | *** join/#asterisk pacific (~pacific@store-fw.porchlight.ca) |
| 16:50.46 | *** join/#asterisk lidl (~little@213-140-22-64.fastres.net) |
| 16:51.58 | lidl | hi, i have in sip.conf an extension named [messagenet-out], and in the extensions one like this: exten => _0X.,1,Dial(SIP/${EXTEN}@messagenet-out) |
| 16:52.19 | lidl | but when i dial,it says it cannot find the messagenet-out host.. |
| 16:52.36 | lidl | shouldn't it refer to the messagenet-out section in sip.conf? |
| 16:52.49 | bannerman | zoa: hey, that's nifty |
| 16:53.29 | _solstice_ | Anyone here ever setup a Asterisk server behind a Cisco Router (which nat's the ip) and was able to use a softphone from outside to reach the asterisk server and dial out? |
| 16:55.23 | *** join/#asterisk mrempire (~trefpunt@h71032.upc-h.chello.nl) |
| 16:55.32 | *** join/#asterisk techie (gus@asterisk.horizonte.us) |
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| 17:00.07 | *** join/#asterisk darby_t (~tom@dnn71.neoplus.adsl.tpnet.pl) |
| 17:02.11 | Nugget | <RyJones> Verizon's DHCP server is dropping my lease after 10 minutes of connection, breaking my connectivity. |
| 17:02.15 | Nugget | <RyJones> I wonder why Verizon is doing this. |
| 17:02.17 | Nugget | <RyJones> I set up a script to renew every minute and called tech support |
| 17:02.20 | Nugget | <RyJones> the support guy said my network card is bad. |
| 17:02.22 | Nugget | <RyJones> he then said there must have been a batch of bad network cards, because they had had 70 of these calls in the last hour. |
| 17:02.25 | Nugget | <RyJones> classic tech support call, I should have taped it. |
| 17:02.28 | Nugget | ^ hah |
| 17:02.36 | Makenshi | haha |
| 17:04.02 | Gand_DJ | lol... what a loser |
| 17:04.09 | Gand_DJ | I work for comcast and that is dumb |
| 17:05.34 | Sedorox | hmmmm |
| 17:08.01 | Gand_DJ | ne1 have sipphone setup on * for inbound? |
| 17:08.52 | jaiger | Nugget, I once called SBC for PPP auth problems with Linux hosts. They told me the phone system was out in my state |
| 17:09.19 | jaiger | never minde that I called from an SBC phone |
| 17:09.41 | jaiger | turns out they had changed from plain-text auth to PAP the night before |
| 17:10.59 | Nugget | heh |
| 17:11.39 | MattH | what's the best way to clear records from the asterisk internal database when I delete a sip extension? |
| 17:12.14 | Moonwick | reinstall. |
| 17:12.18 | Nugget | heh |
| 17:12.21 | Moonwick | :) |
| 17:12.28 | MattH | er... |
| 17:12.32 | Moonwick | reload maybe? |
| 17:12.33 | Nugget | what are you referring to when you say "the asteirsk internal database"? |
| 17:12.40 | MattH | like if I do a dbput |
| 17:12.52 | Moonwick | oh, that |
| 17:12.53 | MattH | or database put xxx from the CLI |
| 17:13.03 | MattH | like where does asterisk actually store that database? |
| 17:13.18 | MattH | I guess I could do a deltree every now and then |
| 17:13.38 | *** join/#asterisk ManxPower (~eric@stirprop-s0-0-0-26.ndcr2.datasync.net) |
| 17:13.43 | Nugget | I'm curious, why are you using using dbput as a component of extension management? |
| 17:14.13 | MattH | to do some things like *69 call tracing, and turn call waiting on and off for extensions |
| 17:14.20 | MattH | is there a better way to do it? |
| 17:14.31 | Nugget | I was just curious, not challenging the practice. |
| 17:14.43 | MattH | just asking =) I'm by no means an asterisk guru |
| 17:15.13 | Nugget | I've only used the internal database for storing callerid lookup tables. I'm not really familiar with all its uses. |
| 17:15.15 | Moonwick | neither are we, or we'd have better things to do than hang out on IRC all day |
| 17:15.21 | MattH | lol |
| 17:15.25 | DrWho17 | MattH: you can store them in an external database |
| 17:15.57 | MattH | DrWho17: would I simple make an AGI script that is called when someone, say turns call waiting on.. run and talk tot he mysql database... ? |
| 17:16.05 | MattH | tot he = to a |
| 17:16.15 | DrWho17 | no, there is a special odbc application to store those externally |
| 17:16.21 | Nugget | eww, mysql. :) |
| 17:16.36 | Nugget | mysql is nowhere to be found on my asterisk machines |
| 17:16.54 | Moonwick | nugget even does an rm -r /usr/ports/databases/mysql* out of spite. |
| 17:17.11 | MattH | lol |
| 17:17.25 | MattH | DrWho17: do you know what it is off hand? |
| 17:17.28 | *** join/#asterisk Gh0sty (~Ghosty@ip-81-11-244-12.dsl.scarlet.be) |
| 17:17.48 | DrWho17 | app_dbodbc |
| 17:18.08 | DrWho17 | makes it easier to manipulate these things via an external web interface |
| 17:18.13 | MattH | oh yes much |
| 17:18.15 | MattH | I wasn't aware of that |
| 17:18.23 | MattH | does it come installed by default in the source code? or do I need to add it in? |
| 17:18.25 | Nugget | I've got a berkely db file /var/lib/asterisk/astdb, I presume that's the file. |
| 17:18.43 | DrWho17 | http://www.voip-info.org/tiki-index.php?page=Asterisk%20app_dbodbc |
| 17:18.57 | MattH | much thanks |
| 17:19.07 | MattH | *twiddles thumbs as the voip-info page loads* |
| 17:19.17 | MattH | *still twiddling*.. wow this site is getting slower and slower |
| 17:22.12 | lidl | hi, i have in sip.conf an extension named [messagenet-out], and in the extensions one like this: exten => _0X.,1,Dial(SIP/${EXTEN}@messagenet-out) |
| 17:22.14 | lidl | but when i dial,it says it cannot find the messagenet-out host.. |
| 17:22.17 | lidl | shouldn't it refer to the messagenet-out section in sip.conf? |
| 17:22.26 | lidl | where the host is defined instead |
| 17:24.24 | emrah | Hello again all. Anyone her is using safe_asterisk? There is a bug in the lasts cvs versions, do you have an idea to repare it? Is it ok to use maybe another version? |
| 17:29.18 | *** join/#asterisk Wazb (Wazb@207.245.215.111) |
| 17:29.21 | Wazb | hi all |
| 17:30.07 | Sedorox | lidl: no... it'll refer to the messagenet-out in extentions.conf |
| 17:31.43 | *** join/#asterisk Romik (~romik@1.fix.netvision.net.il) |
| 17:31.44 | lidl | Sedorox, thx |
| 17:31.53 | *** join/#asterisk ramtha (~tk@td9091901.pool.terralink.de) |
| 17:32.00 | ramtha | hi |
| 17:32.16 | lidl | so how do i define to use all the options i specify in sip.conf using the dial cmd? |
| 17:32.19 | ramtha | i have a TE410P and i have following dialplan |
| 17:32.44 | lidl | for example i'd like to say nat=yes, which is specified in messegenet-out section in sip.conf |
| 17:33.00 | ramtha | exten => _X.,1,Dial(Zap/g1/${EXTEN}),60 |
| 17:33.06 | ramtha | exten => _X.,1,Dial(Zap/g2/${EXTEN}),60 |
| 17:33.09 | ramtha | and so on |
| 17:33.32 | ramtha | exten => _X.,2,Dial(Zap/g2/${EXTEN}),60 |
| 17:33.40 | ramtha | 2 not 1 |
| 17:33.48 | emrah | lastlog emrah |
| 17:33.52 | emrah | sorry |
| 17:33.54 | ramtha | ist there a nicer soulution |
| 17:34.14 | ramtha | for "load balancing) for the spans? |
| 17:34.37 | Sedorox | lidl: whatever sip account that you want to dial out of.. just put it under the messagenet-out |
| 17:35.33 | lidl | Sedorox, but if you see at this page, they use it in the way i was trying to |
| 17:35.39 | lidl | http://www.voipuser.org/forum_topic_1030.html |
| 17:36.07 | lidl | Sedorox, take a look at the sipgate extension |
| 17:36.14 | lidl | exten => _9.,1,Dial(SIP/${EXTEN:1}@sipgate,30,tr) |
| 17:36.24 | lidl | where sipgate is defined in the sip.conf |
| 17:36.25 | Gand_DJ | hrm, trying to park a call but it doesn't seem to be working. :( |
| 17:36.36 | Gand_DJ | dialing ext 700 with open line |
| 17:36.59 | *** join/#asterisk r0d3nt (~RatMan@wsip-24-234-241-84.lv.lv.cox.net) |
| 17:37.31 | Sedorox | so the messagenet-out is the user... |
| 17:37.41 | Sedorox | just use dial(SIP/messagenet-out) |
| 17:38.12 | lidl | Sedorox, is the voip provider i should route calls to |
| 17:38.35 | Sedorox | yea... |
| 17:38.37 | Sedorox | so you wanna call it like |
| 17:38.55 | Sedorox | exten _00.,1,Dial(SIP/messagenet-out,30,rt) |
| 17:38.59 | Wazb | i am trying to install G729 from http://www.readytechnology.co.uk/open/g729/INSTALL-041103.txt , i am confuse in Step 1b |
| 17:39.25 | lidl | Sedorox, without specifying the extension to effectively call? |
| 17:39.26 | Wazb | apt-get build-dep asterisk, where i have to type this command |
| 17:39.38 | Sedorox | Ok |
| 17:39.41 | Sedorox | well then it would be... |
| 17:39.53 | emrah | Gand_DJ: To park a call, you must have an open channel with the T or t paramter, and when you want to parc, press # |
| 17:39.57 | Sedorox | exten _00.,1,Dial(SIP/messagenet-out/${EXTEN:2},30,rt) |
| 17:39.59 | emrah | then enter the parcking extention |
| 17:40.02 | emrah | like 700 |
| 17:40.05 | Sedorox | so its the user/extention |
| 17:40.13 | *** join/#asterisk rg1 (~rg1@mail.airlinksystems.com) |
| 17:40.17 | Sedorox | and thew above example would cut the 00 off the extention |
| 17:40.18 | emrah | you will hear the number wher you can rescupe your call back |
| 17:40.36 | rg1 | has anyone in here used sphinx voice recog on asterisk? |
| 17:41.01 | lidl | *CLI> -- Executing Dial("OSS/dsp", "SIP/messagenet-out/0236522076") in new stack |
| 17:41.02 | lidl | Use EXIT or QUIT to exit the asterisk console |
| 17:41.02 | lidl | *CLI> Apr 21 19:40:54 WARNING[16686]: chan_sip.c:1399 create_addr: No such host: messagenet-out |
| 17:41.03 | *** join/#asterisk zeedo (~zeedo@www.bsrf.org.uk) |
| 17:41.27 | emrah | hum |
| 17:41.37 | Sedorox | you have to have [messagenet-out] in sip.conf with all the info in it.. like user, pass, host... |
| 17:41.37 | emrah | lidl: |
| 17:41.51 | emrah | lidl: What provider would you like to use? |
| 17:42.35 | lidl | [messagenet-out] |
| 17:42.35 | lidl | type=peer |
| 17:42.35 | lidl | secret=secret |
| 17:42.35 | lidl | username=user |
| 17:42.35 | lidl | fromuser=user |
| 17:42.36 | lidl | host=sip.messagenet.it:5061 |
| 17:42.38 | lidl | context=messegenet-in |
| 17:42.40 | lidl | canreinvite=no |
| 17:42.46 | lidl | nat=yes |
| 17:42.52 | Sedorox | ~pastebin |
| 17:42.53 | jbot | well, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 17:43.21 | lidl | Sedorox, sorry, i thought 10 lines could be ok |
| 17:43.22 | Sedorox | change type to be type |
| 17:43.25 | Sedorox | errr |
| 17:43.28 | Sedorox | type to be friend |
| 17:43.30 | *** join/#asterisk rue_mohr (~dan@d154-20-50-233.bchsia.telus.net) |
| 17:43.34 | Sedorox | lidl: its ok |
| 17:43.42 | Sedorox | some people are picky about it |
| 17:43.45 | Sedorox | I personally dun care |
| 17:43.45 | Sedorox | lol |
| 17:43.51 | rue_mohr | who can help me, main guy is on vaccation, asterisk dropping calls all over the place |
| 17:43.55 | *** join/#asterisk gbdrbob (drbob@alltalk.demon.co.uk) |
| 17:44.09 | rue_mohr | logs say alot of things that look evil to me |
| 17:44.13 | rue_mohr | no experiance |
| 17:44.19 | lidl | Sedorox, with type=friend or user is the same as with peer |
| 17:44.51 | lidl | emrah, the provider is in the host |
| 17:44.53 | Sedorox | friend is peer and user |
| 17:45.10 | Sedorox | you want it if you want to be able to use it to dial from.. and recieve from |
| 17:45.12 | lidl | the problem is it seems it doens't see the messagenet-out section |
| 17:45.19 | *** join/#asterisk Fanguin (~Fanguin@p508192B6.dip0.t-ipconnect.de) |
| 17:45.20 | lidl | Sedorox, ok |
| 17:45.25 | rue_mohr | can anyone tell me what I might be looking for? |
| 17:45.51 | rue_mohr | I have a lot of angry people.... |
| 17:46.15 | lidl | <PROTECTED> |
| 17:46.16 | Sedorox | whats on the console? |
| 17:46.21 | bjohnson | rue_mohr: log into the asterisk box |
| 17:46.28 | bjohnson | rue_mohr: then run asterisk -r |
| 17:46.28 | rue_mohr | yup, I'm there |
| 17:46.33 | lidl | Sedorox, may i paste it in a query with you? |
| 17:46.37 | bjohnson | then run set verbose 5 |
| 17:46.41 | Fanguin | hi, did anybody use sems (sip express media server) http://sems.berlios.de/? I would like to know if there are things that sems can do but that cannot be done by asterisk. |
| 17:46.43 | Sedorox | lidl: yes |
| 17:46.43 | rue_mohr | looking through /var/log/asterisk/all |
| 17:46.51 | bjohnson | then make a call that drops and pastebin the messages |
| 17:46.56 | bjohnson | ~pastebin |
| 17:46.57 | jbot | i guess pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca |
| 17:47.13 | rue_mohr | bjohnson ok, hold on... |
| 17:47.18 | bjohnson | rue_mohr: you need to narrow down the messages to one that has a problem |
| 17:47.30 | bjohnson | err .. one call that has a problem |
| 17:47.36 | bjohnson | otherwise too much info |
| 17:48.02 | rue_mohr | bjohnson |
| 17:48.18 | bjohnson | Fanguin: if it's like ser, it may help handle more calls in a high use environment |
| 17:48.52 | rue_mohr | bjohnson I cant make a call that just dropps |
| 17:49.15 | rue_mohr | there dropping randomly |
| 17:49.28 | Sedorox | bandwidth? |
| 17:49.34 | rue_mohr | its fine |
| 17:49.44 | Sedorox | hmmm |
| 17:49.46 | bjohnson | cpu? |
| 17:49.48 | bjohnson | memory? |
| 17:49.55 | rue_mohr | all fine... |
| 17:50.09 | rue_mohr | just a sec... let me get a suspicious entry.. |
| 17:50.26 | Fanguin | bjohnson, no, it's not like ser. the "one sentence description" looks like: The Sip Express Media Server (SEMS) provides audio centric MRF core functionalities like announcements, voice mail and audio conferencing. I found this description on |
| 17:50.28 | Fanguin | http://www.fokus.gmd.de/bereichsseiten/testbeds/ims_playground/playground/media_Server.php?lang=de |
| 17:50.50 | bjohnson | not sure if we can help with random dropped calls without some kind of log messages |
| 17:51.21 | Fanguin | bjohnson, sems must be used together with ser. sems can be found here: http://sems.berlios.de/ |
| 17:51.27 | bjohnson | rue_mohr: describe your voip system (ie fxs, fxo, PRI, voip phones, etc) |
| 17:51.35 | rue_mohr | Didn't get a frame from channel: Zap/2-1 |
| 17:52.09 | bjohnson | Fanguin: perhaps something that operates similarly to * then .. but sip only? |
| 17:52.32 | rue_mohr | oh, dear do I know that much, its a Norstar system with asterisk connected via T1 routing calls to other asterisk maches to other norstars |
| 17:52.48 | rue_mohr | ? |
| 17:52.53 | *** join/#asterisk Juxt (~Juxt@64.135.20.202) |
| 17:53.07 | rue_mohr | sorry, I'm trying to hold back ther users here while I deal with it |
| 17:53.20 | bjohnson | so N<->*<->internet<->*<->N ? |
| 17:53.29 | bjohnson | and it's the internet calls that are dropping? |
| 17:53.30 | Fanguin | bjohnson, yes perhaps ... :-) |
| 17:53.31 | Wazb | i am trying to install G729 fom intel website, where i have to type this command apt-get build-dep asterisk , help please! |
| 17:53.37 | bjohnson | ie pstn calls are fine? |
| 17:53.48 | rue_mohr | its a private network |
| 17:53.52 | bjohnson | rue_mohr: put them one |
| 17:53.54 | bjohnson | err |
| 17:53.56 | bjohnson | rue_mohr: put them on |
| 17:54.34 | bjohnson | rue_mohr: multiple nortels connected with asterisk machines on a private lan? |
| 17:54.35 | Juxt | hello |
| 17:54.39 | rue_mohr | :) their just users, to them, the phone system is dropping calls and thats all they know and there just angry |
| 17:54.40 | bjohnson | rue_mohr: big system is it |
| 17:54.41 | rue_mohr | :) |
| 17:54.47 | Juxt | can someone clarify how accountcode is used in cdr billing? |
| 17:54.48 | rue_mohr | yes |
| 17:54.53 | rue_mohr | inter office |
| 17:55.03 | rue_mohr | 3 sites |
| 17:55.14 | bjohnson | Juxt: accountcode is a text string that you can set that gets listed on each line of the cdr |
| 17:55.24 | Juxt | ok that's what i thought |
| 17:55.26 | Juxt | cool thank you |
| 17:55.34 | rue_mohr | about 100 or so phones... |
| 17:55.43 | bjohnson | useful for tagging which calls go out which voip provider for instance |
| 17:56.09 | Juxt | yeah i was using it to tag billing for my hosted pbx stuff |
| 17:56.11 | rue_mohr | heh, no normal provider bridges here, too slow and chunkey |
| 17:56.18 | Juxt | but i wasn't sure it was used anywhere else |
| 17:56.20 | bjohnson | rue_mohr: one system has direct pstn access? |
| 17:56.41 | rue_mohr | yes, via T1 |
| 17:56.47 | *** join/#asterisk Skarmeth (~Skarmeth@201009017044.user.veloxzone.com.br) |
| 17:57.16 | bjohnson | so N<->*<->telco T1 for that system? |
| 17:57.19 | Juxt | is there a way to add additional variables to the cdr record? |
| 17:57.34 | rue_mohr | bjohnson yes |
| 17:57.39 | rue_mohr | bjohnson just a sec... |
| 17:57.56 | bjohnson | rue_mohr: do phones on that particular Nortel drop calls too? |
| 17:57.59 | Elshar | Hey, I have a simple question about outward calling.. was trying to find the answer on the xvoip forums, but they seem to be down again.. |
| 17:58.03 | *** part/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca) |
| 17:58.15 | bjohnson | Juxt: I think you can if using a database |
| 17:58.23 | Juxt | i am using a database |
| 17:58.27 | *** join/#asterisk djin (~djin@gridfox.xs4all.nl) |
| 17:58.28 | bjohnson | Juxt: but I don't really know |
| 17:58.33 | Juxt | ok |
| 17:58.36 | Juxt | i'll keep looking |
| 17:58.37 | Elshar | If I make an extension, like _9XXX,1,dial(somestuff), and then after do _9XXX,2,dial(somethingelse), it should try 1 and then 2 right? |
| 17:58.41 | Elshar | Or did I miss something? |
| 17:58.48 | bjohnson | you missed it |
| 17:59.02 | rue_mohr | hmm,m ok, we have two asterisk boxes here... |
| 17:59.05 | bjohnson | if priority 1 dial() works then 2 never runs |
| 17:59.17 | rue_mohr | one just be telco, other must be local...??? |
| 17:59.20 | bjohnson | if priority 1 dial() gets busy, then it jumps to 102 |
| 17:59.22 | rue_mohr | hmm |
| 17:59.43 | rue_mohr | bjohnson its one site thats having trouble |
| 18:00.04 | bjohnson | rue_mohr: trying to narrow down the problem .. find the shortest route to the telco and see if it is dropping calls |
| 18:00.28 | bjohnson | rue_mohr: you can confirm that it is one site that is connected to the main site? |
| 18:00.43 | bjohnson | Elshar: look into the superdial macro on the wiki |
| 18:00.59 | Elshar | superdial? Will do, thanks for the heads up. :) |
| 18:01.57 | rue_mohr | bjohnson its definitly 1 site, I'm here at the main site... |
| 18:02.10 | rue_mohr | IAX2/astpbx-pstn/16386 stopped sounds |
| 18:02.14 | rue_mohr | that bad? |
| 18:02.24 | rue_mohr | Goto (macro-out-pstn,s-pstnoverflow,1) |
| 18:02.28 | rue_mohr | overflow!? |
| 18:02.32 | bjohnson | rue_mohr: pastebin all the log messages related to that call |
| 18:02.40 | rue_mohr | ok |
| 18:03.00 | bjohnson | and grep the log for other occurences of that macro getting called |
| 18:03.01 | Skarmeth | does cvs.digium.com it's off-line? |
| 18:03.07 | Dutts | anyon ehere using the TE110P int he UK/ |
| 18:03.10 | Micc | SIP port is 5060 UDP, right? |
| 18:03.10 | Qwell | Skarmeth: Its a round robin DNS, try again |
| 18:03.11 | bjohnson | and pastebin that macro from /etc/asterisk/extensions.conf |
| 18:03.22 | bjohnson | Micc: that's one of them |
| 18:03.36 | rue_mohr | http://www.pastebin.com/274191 |
| 18:03.43 | bjohnson | Micc: SIP also uses 10000-20000 by default for the rtp streams |
| 18:04.26 | *** join/#asterisk Blackthorn (blackthorn@ws-10.smyth.net) |
| 18:04.45 | Blackthorn | Hello, do you know how to disable call-waiting on a sipura ata? |
| 18:05.38 | *** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc) |
| 18:06.16 | rue_mohr | 'Zap/23-1' seems to be related to those overflows, what is it? |
| 18:06.59 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.res.rr.com) |
| 18:08.09 | *** join/#asterisk loick (~loick@APuteaux-151-1-47-42.w82-124.abo.wanadoo.fr) |
| 18:10.10 | rue_mohr | bjohnson ? |
| 18:10.50 | Sedorox | ~wiki-status |
| 18:10.51 | jbot | wiki-status is probably Up and Running |
| 18:11.02 | Sedorox | jbot: no, wiki-status is Slow like normal |
| 18:11.03 | jbot | okay, Sedorox |
| 18:11.24 | rue_mohr | sorry for my lack of experiance here |
| 18:11.29 | Qwell | Sedorox: up and walking? |
| 18:11.41 | Qwell | or crawling |
| 18:12.10 | rue_mohr | there's another thing here, I cant find it now, something about duplicate answered calls... |
| 18:12.52 | rue_mohr | DEBUG[8155]: Dropping duplicate answer! |
| 18:13.00 | UBiQUiTY | i need an agi script to dial "1#", then delay 1 second, and loop indefinately until hangup ... I havent written an agi script before, and I'm sure this would be rather easy. can anyone help? |
| 18:13.03 | Sedorox | lol |
| 18:13.12 | Sedorox | actually.. first load was slow.. isn't too bad now |
| 18:14.58 | *** join/#asterisk gpowers (~glenn@static-68-162-84-101.phil.east.verizon.net) |
| 18:16.22 | rue_mohr | http://www.pastebin.com/274204 I dont know whats actaully bad just what soudns bad... |
| 18:17.08 | rue_mohr | bjohnson you still here? |
| 18:17.38 | UBiQUiTY | can an agi be written in shell script? |
| 18:17.53 | MattH | DrWho17: are you still here? |
| 18:18.22 | MattH | DrWho17: do you have any idea what is ment by the line app_setcdruserfield.so app_random.so app_dbodbc.so at http://www.voip-info.org/tiki-index.php?page=Asterisk%20app_dbodbc am I to replace that line with what is there? |
| 18:19.26 | rue_mohr | what are typical causes of dropps? |
| 18:19.44 | rue_mohr | bjohnson are you waiting for a macro from me? |
| 18:20.58 | Micc | is asterisk win32 0.52 known to work with sip? I'm getting connection reset by peer errors. |
| 18:21.29 | rue_mohr | bjohnson http://www.pastebin.com/274206 that? |
| 18:21.44 | *** join/#asterisk Borgon (~Borgon@vl135-238.vl135.GeorgiaSouthern.edu) |
| 18:21.47 | rue_mohr | I dont even know whats part of the default config or not |
| 18:22.27 | rue_mohr | <PROTECTED> |
| 18:22.27 | rue_mohr | <PROTECTED> |
| 18:22.27 | rue_mohr | <PROTECTED> |
| 18:22.27 | rue_mohr | <PROTECTED> |
| 18:22.27 | rue_mohr | <PROTECTED> |
| 18:22.29 | rue_mohr | <PROTECTED> |
| 18:22.31 | rue_mohr | <PROTECTED> |
| 18:22.33 | rue_mohr | <PROTECTED> |
| 18:22.34 | Dutts | anyone here operating a voip gateway? I'm interested in finding out how many simultaneous calls you can fit down a 512/256 adsl connection? I knwo it depends on the codec but anyone got any figures? |
| 18:22.35 | rue_mohr | <PROTECTED> |
| 18:22.37 | rue_mohr | Apr 21 11:22:18 WARNING[8331]: app_dial.c:362 wait_for_answer: Unable to forward frame |
| 18:22.39 | rue_mohr | Apr 21 11:22:18 WARNING[8331]: app_dial.c:362 wait_for_answer: Unable to forward frame |
| 18:22.40 | tzanger | rue_mohr: stop flooding |
| 18:22.41 | rue_mohr | oh man sosrry!!!! |
| 18:22.49 | Borgon | hello.. can i use asterisk with a regular voip provider and a headset? i have a ethernet card on a edu college lan.. will it work? |
| 18:22.58 | rue_mohr | really sorry, forgot to hit ctrl-c on teh one line I wanted to paste there |
| 18:23.29 | bjohnson | Blackthorn: under cwi somewhere in the web setup |
| 18:23.59 | bjohnson | rue_mohr: zap/23 is a PRI channel .. likely a fxs in this case |
| 18:24.28 | rue_mohr | bjohnson ok... |
| 18:24.35 | bjohnson | looks like this asterisk server is forwarding the call to another server (astpbx-pstn at 10.0.55.246) |
| 18:24.55 | rue_mohr | yes |
| 18:25.07 | rue_mohr | let me check that ip against what I know |
| 18:25.22 | bjohnson | Borgon: yes but you don't even need * for that |
| 18:25.40 | bjohnson | it's being accepted |
| 18:26.09 | bjohnson | see if you can find the log file related to this particular incoming call .. I bet it is then dialing out another zap channel |
| 18:26.15 | rue_mohr | bjohnson ok, hold on while I figure out what that ip is... |
| 18:26.35 | bjohnson | are you certain that this call is one that was dropped? |
| 18:26.41 | rue_mohr | no |
| 18:26.54 | rue_mohr | there are approx 1700lines/6min going by here |
| 18:27.12 | bjohnson | so far the logs from this machine look like you have an overly complicated dialplan .. but that it works as expected |
| 18:27.35 | rue_mohr | ok :) |
| 18:27.57 | bjohnson | so we need to check the logs from the other machine related to this particular call |
| 18:28.06 | rue_mohr | I have to find it |
| 18:28.13 | rue_mohr | more reverse engineering of the network... |
| 18:28.43 | UBiQUiTY | does anyone know how to send DTMF signals from within AGI ?> |
| 18:28.48 | rue_mohr | 10. is the phone dedicated network... |
| 18:29.34 | Micc | Anyone here know much about the win32 port of asterisk? |
| 18:30.02 | ChkDigit | I know enough to say "Why?" |
| 18:30.26 | ChkDigit | Nothing like taking down an entire PBX because of Windows Update. |
| 18:30.59 | Borgon | hello.. can i use asterisk with a regular voip provider and a headset? i have a ethernet card on a edu college lan.. will it work? |
| 18:31.07 | Sedorox | lol |
| 18:31.20 | Sedorox | Borgon: why not... |
| 18:31.44 | Sedorox | works for me.. I jjust had to ask them to put the port in the PIX so it had a higher priority |
| 18:31.46 | *** join/#asterisk pointer (pointer@aj.catt.com) |
| 18:31.56 | ChkDigit | Borgon: Yes, but that appears to be overkill... |
| 18:32.20 | pointer | anyone know what the deal is with nufone's upgrades? |
| 18:32.24 | Micc | ChkDigit, I installed it and copied my sip.conf from my linux box and started it up and I'm getting recv read errors. |
| 18:33.14 | Micc | ChkDigit, chan_sip.cA:7762 sipsock_read: Recv error: Conneciton reset by peer |
| 18:34.46 | *** part/#asterisk pointer (pointer@aj.catt.com) |
| 18:34.48 | ChkDigit | Micc: Sounds like the remote is not accepting connections on that port. |
| 18:35.14 | bjohnson | Borgon: same answer as before |
| 18:35.29 | tzanger | bkw_: can you reset bugs2 passwords? |
| 18:35.32 | bjohnson | Borgon: yes but you don't even need * for that |
| 18:35.42 | Micc | ChkDigit, I would think that too, but I've seen this error a lot before when porting. |
| 18:37.36 | *** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com) |
| 18:39.46 | ChkDigit | Micc: That would typically be from a failure in connect() |
| 18:39.50 | *** join/#asterisk vaxen (~vaxen@82-70-153-250.dsl.in-addr.zen.co.uk) |
| 18:40.03 | Juxt | is it possible to use a voicemodem with firefly? |
| 18:40.17 | bjohnson | Juxt: no |
| 18:40.32 | *** join/#asterisk Blackthorn (blackthorn@ws-10.smyth.net) |
| 18:41.37 | Blackthorn | I have a sipura ata behind nat, it has a static ip, and the router port forwards 5060,5061, 10000, and 16384. Should I also need to have nat keep alive set to on? |
| 18:42.40 | bjohnson | yes |
| 18:42.51 | jabbzy | yes |
| 18:43.02 | bjohnson | is * behind another nat too? |
| 18:43.27 | *** join/#asterisk MajestiK (~MajestiK@S01060800208687ec.ed.shawcable.net) |
| 18:43.27 | Blackthorn | no |
| 18:43.36 | Blackthorn | * is on routable. |
| 18:43.41 | bjohnson | also I don't think you need to forward the ports on the router providing NAT to the SPA |
| 18:44.18 | bjohnson | I think add nat=yes in sip.conf and you should be ok |
| 18:44.34 | Blackthorn | well.. i've tried it without forwarding on several of my installs and I can't get connections. But perhaps because keep alive is no on? nat is set to yes however. |
| 18:44.53 | bjohnson | keep alive keeps the connections alive |
| 18:45.12 | Blackthorn | ok, i'll give it a try and see what happens. |
| 18:45.14 | bjohnson | I don't think you need th ports forwarded .. and if you do, it's 10000-20000 |
| 18:45.40 | Micc | ChkDigit, yeah I would say so too but sip show registry shows that I'm registered. |
| 18:45.59 | PTG1234 | yah no port forwarding needed |
| 18:46.06 | PTG1234 | but set reinvite on the device to 120seconds |
| 18:46.10 | PTG1234 | and qualify=yes on the asterisk box |
| 18:46.39 | *** join/#asterisk oej (~oej@206.129.72.6) |
| 18:47.18 | Borgon | bjohnson: i know, but i was asking if i needed some kind of other switch or hardware |
| 18:47.45 | Blackthorn | what does qualify=yes mean? |
| 18:47.59 | rue_mohr | bjohnson I found the comptuer! |
| 18:48.02 | *** part/#asterisk Romik (~romik@1.fix.netvision.net.il) |
| 18:48.18 | Sedorox | sip/iax2 show peers/users, will show the latency |
| 18:49.04 | PTG1234 | it is somethingy ou need when going through firewalls |
| 18:49.33 | *** join/#asterisk L|NUX (~linux@202.5.145.58) |
| 18:50.57 | *** join/#asterisk brad[] (~brad@brad.developer.gentoo) |
| 18:51.00 | brad[] | Hello folks |
| 18:51.02 | Borgon | is there a free voip provider like pruvider, that gives iax control to work with astreriks? |
| 18:51.12 | lidl | why asterisk could ignore a section which exists in sip.conf when i specify it in a dial command? |
| 18:51.26 | brad[] | I have a SIP phone connected to asterisk and configured on both ends to pass dtmf along as SIP INFO, but get this error: |
| 18:51.27 | brad[] | Apr 21 10:52:12 WARNING[9730]: chan_sip.c:6134 receive_info: Unable to parse INFO message from 706b1d591e486a5326fda1ec0e99cabf@192.168.0.208. Content |
| 18:51.39 | brad[] | Where do I begin looking after this? |
| 18:51.41 | lidl | the sip configuration is at: http://pastebin.ca/10018 |
| 18:52.05 | Blackthorn | the sip phone allws you to set the nat notify that is sent to *.. should it be the default notify or blank? |
| 18:52.13 | lidl | the dial command is: exten => _0X.,1,Dial(SIP/messagenet-out/${EXTEN}) |
| 18:52.14 | Blackthorn | I read in the instructions it can be either. |
| 18:54.32 | Juxt | how do a set a caller id for the whole context? |
| 18:55.14 | *** join/#asterisk Bile_One (~bile_one@pcp03281999pcs.gillst01.ar.comcast.net) |
| 18:55.31 | bjohnson | Borgon: do you mean like fwd? |
| 18:55.33 | bjohnson | ~fwd |
| 18:55.34 | jbot | from memory, fwd is Free World Dialup: Brainchild of Jeff Pulver. URL: http://www.pulver.com/fwd/ |
| 18:55.47 | bjohnson | rue_mohr: hehe .. now find a dropped call in the logs |
| 18:56.12 | bjohnson | rue_mohr: easiest if a user can give you a phone number that matches up with a dropped call |
| 18:56.22 | *** join/#asterisk Veryhot (~tho@adsl-69-109-159-239.dsl.sndg02.pacbell.net) |
| 18:56.31 | bjohnson | rue_mohr: make sure you find out if they called it multiple times and which one was dropped |
| 18:56.49 | bjohnson | rue_mohr: such a user should be easy to find .. pick one that is complaining to you |
| 18:57.06 | *** join/#asterisk Xen^ (~linux@202.5.145.58) |
| 18:58.28 | bjohnson | lidl: I thought SIP dial commands were more like dial(SIP/${EXTEN}@server) .. or are you trying to use a dial where a goto should be used? |
| 18:59.10 | bjohnson | Juxt: setcallerid() in extensions.conf |
| 18:59.53 | rue_mohr | most of them are fed up with complaining now... :/ |
| 19:00.01 | rue_mohr | I'll dig a bit here |
| 19:00.02 | Bile_One | Anyone in here know about using call files to call one asterisk via IAX2 trunking? |
| 19:00.25 | Borgon | bjohnson: i want to have control of the iaxp |
| 19:01.55 | marlowe | anyone using livevoip |
| 19:02.51 | Borgon | do free voip provider give you access to iax? |
| 19:03.03 | *** join/#asterisk elriah (~jfulcrum@adsl-068-209-198-242.sip.bhm.bellsouth.net) |
| 19:03.12 | *** join/#asterisk ramtha (~tk@td9091901.pool.terralink.de) |
| 19:03.20 | *** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co) |
| 19:03.21 | elriah | Hi all, is there an official term for a Voice Over IP PBX? Or is it still just a PBX? |
| 19:03.48 | Bile_One | yea it is called Asterisk |
| 19:03.59 | Makenshi | no it isn't |
| 19:04.03 | ramtha | hm i have two gateway configured in iax.conf. but i can only authenticate with the last one in config file. why is that? |
| 19:04.06 | Bile_One | :P |
| 19:04.11 | Makenshi | Our Siemens iSDX switches aren't asterisk :P |
| 19:04.21 | Bile_One | I was joking |
| 19:04.24 | *** part/#asterisk Fanguin (~Fanguin@p508192B6.dip0.t-ipconnect.de) |
| 19:04.41 | ramtha | neo metter how much i out in iax.conf, only the last one works |
| 19:04.57 | *** join/#asterisk L|NUX (~linux@202.5.145.58) |
| 19:05.00 | Borgon | do free voip provider give you access to iax? |
| 19:05.04 | *** join/#asterisk NormAst (~NormAst@CPE000625ee7e4e-CM0012c90d3496.cpe.net.cable.rogers.com) |
| 19:05.11 | NormAst | Hi all. |
| 19:05.21 | AgiNamu | Borgon, what's a "free voip provider"? |
| 19:05.25 | AgiNamu | hi |
| 19:05.34 | ramtha | Borgon: i donīt think so |
| 19:05.38 | bjohnson | Borgon: fwd does .. I have no idea what iaxp means |
| 19:06.07 | Borgon | bjohnson: what is fwd ? iaxp sorry i mean iax, so i can use asterisk with it |
| 19:06.11 | bjohnson | ramtha: it means you've configured then incorrectly |
| 19:06.19 | bjohnson | well .. again |
| 19:06.21 | bjohnson | ~fwd |
| 19:06.22 | jbot | i guess fwd is Free World Dialup: Brainchild of Jeff Pulver. URL: http://www.pulver.com/fwd/ |
| 19:06.40 | bjohnson | seems I'm having to say everything twice to you |
| 19:06.44 | bjohnson | so .. |
| 19:06.45 | bjohnson | ~fwd |
| 19:06.46 | jbot | rumour has it, fwd is Free World Dialup: Brainchild of Jeff Pulver. URL: http://www.pulver.com/fwd/ |
| 19:06.47 | ramtha | bjohnson: hmm i have testet every user sektion stand alone and they work |
| 19:07.04 | ramtha | if i put them zogether in one file only the last one works |
| 19:07.07 | brad[] | How would I begin troubleshooting this? : Apr 21 10:52:12 WARNING[9730]: chan_sip.c:6134 receive_info: Unable to parse INFO message from 706b1d591e486a5326fda1ec0e99cabf@192.168.0.208. Content |
| 19:07.12 | ramtha | what is wrong there? |
| 19:07.14 | brad[] | That's on a hook flash |
| 19:07.37 | lidl | bjohnson, look at http://www.voipuser.org/forum_topic_1030.html |
| 19:07.48 | lidl | take a look at the dial command referring to sipgate |
| 19:07.53 | bjohnson | ramtha: likely you should read the wiki page about iax.conf authentication |
| 19:07.56 | lidl | and then look where sipgrate is |
| 19:08.02 | lidl | in the sip.conf |
| 19:08.23 | lidl | i'd like to use asterisk in the same way this page shows |
| 19:08.38 | bjohnson | brad[]: didn't you say you were using dtmf in INFO mode |
| 19:08.57 | brad[] | bjohnson: Yeah, on both the SIP phone and in sip.conf |
| 19:09.03 | brad[] | For that particular phone |
| 19:09.17 | bjohnson | brad[]: I'd start by changing the dtmf mode to something else .. like rfc2833 |
| 19:09.28 | brad[] | Tried, not working |
| 19:09.46 | brad[] | INFO brought me closest to success, but I don't know why asterisk wouldn't know what the SIP phone was sending it |
| 19:09.56 | bjohnson | no idea |
| 19:10.03 | bjohnson | I use rfc2833 for everything |
| 19:10.26 | brad[] | bjohnson: Do you have SIP phones? |
| 19:10.39 | bjohnson | SIP ATAs |
| 19:10.56 | brad[] | bjohnson: What kind? |
| 19:11.05 | bjohnson | 3 SPA 2000 and 3 SPA 3000 |
| 19:11.34 | brad[] | bjohnson: Excellent - what's the equivalent setting for rfc2833 on the ATA? I have INFO and AVT as options for hook flash on my ATA-2000. |
| 19:11.39 | brad[] | sorry SPA-2000. |
| 19:11.54 | johnnyb | How does fwd make money? |
| 19:12.21 | bjohnson | lidl: you have * behind a nat router? |
| 19:12.35 | bjohnson | johnnyb: how do you make money? |
| 19:12.44 | bjohnson | johnnyb: and can I have some? |
| 19:12.48 | lidl | bjohnson, behind nat, yes |
| 19:13.02 | Borgon | whats the disadvantage of using fwd? |
| 19:13.02 | Borgon | bjohnson: they seem to have affiliates.. peer 1 and others |
| 19:13.16 | bjohnson | lidl: sipgate will likely have some setup info .. running sip through nat is always a pita |
| 19:13.16 | AgiNamu | ...I want to use NAT VOIP IAX RTP to save money on VOIP NANPA CLEC RBOC TRAFFIC... |
| 19:13.50 | AgiNamu | Do the free PSAP interconnect providers support CORBA? |
| 19:13.56 | lidl | bjohnson, mine is a syntax problem 1st of all |
| 19:14.04 | bjohnson | Borgon: you'll have to figure out 1. what you want and 2. who will give it to you for free. I'm not going to evaluate every service provider's offerings |
| 19:14.19 | lidl | bjohnson, i receive calls regurarly |
| 19:14.29 | Borgon | ok ok |
| 19:14.32 | bjohnson | brad[]: AVT |
| 19:14.37 | brad[] | bjohnson: Excellent |
| 19:14.46 | lidl | bye, i'll og |
| 19:14.48 | lidl | i'll go |
| 19:14.52 | Borgon | bjohnson: wll i just want to spoof my ani |
| 19:14.53 | bjohnson | brad[]: make a note in your sip.conf so you don't forget |
| 19:15.13 | AgiNamu | Borgon, then get a PRI and a provider that allows you to do that. |
| 19:15.23 | AgiNamu | Or find a traffic terminator that allows that. |
| 19:15.29 | brad[] | bjohnson: Thanks much |
| 19:15.43 | AgiNamu | But you wont get that for free. NEXT. |
| 19:15.50 | Borgon | since voip is over the internet.. is it possible to have a proxy or some layer of protection? |
| 19:15.50 | Borgon | AgiNamu: thats too expensive, am in college |
| 19:15.50 | Borgon | AgiNamu: as long as i can get the anac to read my good cpn then ill be happy |
| 19:16.11 | AgiNamu | then find a termination provider that lets you send your own ANI |
| 19:16.18 | Borgon | AgiNamu: well as long as the cpn gets spoof am fine, dont need the ani |
| 19:16.21 | AgiNamu | there's this cool thing that find them |
| 19:16.26 | AgiNamu | i forget what its called |
| 19:16.28 | AgiNamu | oh wait, google. yea |
| 19:16.31 | AgiNamu | i think it's www.google.net |
| 19:16.32 | Borgon | termination provider? |
| 19:16.34 | AgiNamu | something like that. |
| 19:16.45 | AgiNamu | yes, someone who terminates your calls. |
| 19:17.04 | AgiNamu | ~google voip termination |
| 19:17.25 | AgiNamu | that should get you started. enjoy |
| 19:17.42 | Borgon | ok |
| 19:17.42 | Borgon | thank you baby |
| 19:17.51 | AgiNamu | uh yea, ok. |
| 19:20.29 | *** join/#asterisk elawman (~elawman@squid.eastwestp.com) |
| 19:20.34 | bjohnson | AgiNamu <- provides howto info to wannabe phreakers |
| 19:20.41 | AgiNamu | damn straight |
| 19:20.54 | bjohnson | thank you baby |
| 19:21.02 | AgiNamu | i figure if he coudln't figure that out himself, saying "get a PRI" ain't gonna help. |
| 19:21.05 | AgiNamu | or herself. |
| 19:21.12 | elawman | quick question |
| 19:21.26 | elawman | does anyone know the status of using hint priorities in realtime in asterisk cvs-head? |
| 19:23.52 | elawman | I know that as of december, they weren't working |
| 19:24.03 | Borgon | bjohnson: i dont consider myself a phreaker |
| 19:24.06 | AgiNamu | try -dev? ;) |
| 19:24.12 | AgiNamu | Borgon, no, you aren't. |
| 19:24.18 | AgiNamu | He said "wannabe" |
| 19:24.27 | AgiNamu | few orders of magnitude difference |
| 19:25.40 | Juxt | it is normal for DTFM not to work with some voip carriers? |
| 19:25.47 | AgiNamu | Juxt, not normal. |
| 19:26.05 | AgiNamu | it means they aren't handling DTMF correctly (obviously). they should fix it. or you arne't configred like them |
| 19:26.06 | bjohnson | Juxt: some dtmf modes don't work in anything but ulaw |
| 19:26.10 | Borgon | is the windows binary version of asterisk stable? |
| 19:26.10 | Borgon | hehe well am not even a wannabe, am just trying to get free spoofing so covercall can kiss my ars |
| 19:26.15 | Juxt | oh! |
| 19:26.18 | Juxt | i am using ilbc |
| 19:26.24 | AgiNamu | Borgon, forget free. |
| 19:26.35 | AgiNamu | Borgon, the Win32 port is very stable. I hear its even a bit faster too. |
| 19:26.50 | *** join/#asterisk eivindtr (~eivindtr@062016241059.customer.alfanett.no) |
| 19:26.58 | Micc | AgiNamu, I'm having problems with it. |
| 19:27.05 | AgiNamu | with what? |
| 19:27.06 | bjohnson | Juxt: you should google ilbc and dtmf .. I think only rfc2833 works with ilbc |
| 19:27.20 | AgiNamu | oh, using sip?? |
| 19:27.24 | Micc | AgiNamu, the win32 port. yes using sip. |
| 19:27.30 | Micc | Its got read errors. |
| 19:27.31 | AgiNamu | Micc, sarcasm. |
| 19:27.55 | AgiNamu | Like when someone comes in and says "I have a call centre and want to buy unlimited termination for $20 a month" |
| 19:27.59 | *** join/#asterisk L|NUX (~linux@202.5.145.58) |
| 19:28.11 | AgiNamu | or someone comes aroudn and says "I want to call every number in the world and see if it's a fax or an answering machine." |
| 19:28.29 | *** join/#asterisk ast_freak (~ast_freak@hades-out.universalsystems.net) |
| 19:28.35 | AgiNamu | then you reply with extreme sarcasm, hoping they either piss off, or actually believe you and waste time. |
| 19:28.47 | Micc | ok, I gotcha. |
| 19:29.02 | Borgon | ok great, well am in college so i try to save up for my weekend partying hehe |
| 19:29.02 | Borgon | AgiNamu: last question, anyway to be anonymous with voip? do you know of any projects or methods to do this? |
| 19:29.22 | AgiNamu | sure, don't make any calls. |
| 19:29.35 | AgiNamu | Nothing is anonymous against sufficient technology. |
| 19:29.55 | AgiNamu | So if you call up the whitehouse and say you are gonna make shish-ka-bushes, they'll find you. |
| 19:29.59 | file[laptop] | we know your IP... we know your ISP... we can know where you live... |
| 19:30.12 | bjohnson | within minutes usually |
| 19:30.29 | AgiNamu | in fact, I think I'll report your IP just in case |
| 19:31.01 | AgiNamu | Georgia Southern? They just passed that new bill |
| 19:31.08 | AgiNamu | so you'll be double screwed if you try anything. |
| 19:31.16 | *** join/#asterisk mrkyr (~bviitanen@h24-207-80-55.cst.dccnet.com) |
| 19:31.25 | bjohnson | hanging offense? |
| 19:31.34 | AgiNamu | I reckon it might be. |
| 19:31.41 | bjohnson | shucks |
| 19:32.00 | AgiNamu | yea, around these parts, we folks don't take none of that spoofin' nonsense lightly ya know? |
| 19:32.08 | Borgon | lol for sure, always |
| 19:32.08 | Borgon | you go do that |
| 19:32.08 | Borgon | anyways |
| 19:32.08 | Borgon | am gonna go google and try to find some good stuff |
| 19:32.08 | Borgon | georgia southern loves me |
| 19:32.09 | Borgon | dude, am not gonna try anything in the whitehouse |
| 19:32.10 | Borgon | you know why? |
| 19:32.12 | Borgon | because even though i hate the dumbass bush, i dont want dick gayney running the damn country |
| 19:32.13 | Juxt | ugh this dtmf stuff is weird |
| 19:32.35 | *** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode) |
| 19:32.41 | AgiNamu | Juxt, use IAX! :D |
| 19:32.42 | Blackthorn | An person ealier had suggested that i use a command "qualify=yes". Is that command to be put into the sip conext for the specific phone? |
| 19:32.47 | Juxt | i am using iax |
| 19:32.53 | AgiNamu | Blackthorn, yea you can put it in |
| 19:32.57 | Juxt | i am using iax and ilbc as a codec |
| 19:32.58 | eKo1 | Blackthorn: yes |
| 19:33.02 | bjohnson | Blackthorn: or no |
| 19:33.05 | AgiNamu | Juxt, um, then there's only one real DTMF mode: Out of band. |
| 19:33.08 | bjohnson | could be a general thing |
| 19:33.13 | AgiNamu | In fact, Asterisk won't even try it inband, AFAIK |
| 19:33.24 | eKo1 | bjohnson: no, it can't |
| 19:33.25 | AgiNamu | although the IAX2 devices i use, for some reason, can do inband dtmf |
| 19:33.34 | bjohnson | eKo1: damn |
| 19:33.38 | Juxt | what does inband and out-of-band mean |
| 19:33.40 | file[laptop] | not like inband DTMF is hard, 'tis just audio! |
| 19:33.45 | Blackthorn | ok thanks. |
| 19:34.06 | AgiNamu | Inband means it's audio |
| 19:34.06 | file[laptop] | silly silly audio |
| 19:34.06 | AgiNamu | it transmits the tones as compressed audio |
| 19:34.06 | eKo1 | outband means it is sent using sip or whatever |
| 19:34.07 | AgiNamu | and that doesnt work, since most codecs aren't gearted towards tones |
| 19:34.14 | Borgon | i rather have a dumbass running it, than a sick bastard.. i voted for kerry |
| 19:34.14 | Borgon | AgiNamu: well am just trying to get free spoofing than giving my money away to covertcall, gonna prank my mom |
| 19:34.14 | Borgon | AgiNamu: i already can do cpn via vxml.. really easy |
| 19:34.15 | AgiNamu | out-of-band means its send in a control or other packet |
| 19:34.16 | Juxt | right |
| 19:34.23 | ast_freak | I'm looking for some help with Voicemail. I would like to call my own voice mailbox, and instead of leaving a message, press a key and listen to my voicemail. Can someone give me a hand? |
| 19:34.26 | *** part/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com) |
| 19:34.36 | Juxt | so out-of-band should be pretty reliable |
| 19:34.41 | bjohnson | AgiNamu: make your new friend go away |
| 19:34.43 | AgiNamu | out of band is very relliable. |
| 19:34.48 | AgiNamu | op me |
| 19:34.55 | AgiNamu | /kline borgon? |
| 19:34.56 | Juxt | i don't see it being supported by asterisk |
| 19:35.05 | Blackthorn | thanks about the inband /outband.. i had always wondered what that meant as well. |
| 19:35.11 | AgiNamu | Juxt, well, it is.... as I said, that's the only thing that Asterisk will do with IAX |
| 19:35.24 | jabbzy | hey there folks, does anyone know what dialplan.agi does - in a quick summary? |
| 19:35.43 | ast_freak | I read this in the wiki: |
| 19:35.44 | ast_freak | Also. during the prompt if the caller presses: |
| 19:35.46 | ast_freak | <PROTECTED> |
| 19:35.47 | ast_freak | <PROTECTED> |
| 19:35.49 | ast_freak | But it doesn't seem to work right. |
| 19:35.51 | rue_mohr | bjohnson I just found out that everyone was getting dropped at once! |
| 19:35.59 | Juxt | AgiNam i do not see it documented anywhere |
| 19:36.07 | bjohnson | rue_mohr: likely connectivity blip |
| 19:36.27 | rue_mohr | but I'm getting a max of 23ms ping time |
| 19:36.38 | bjohnson | rue_mohr: points to the connection between the local server and the main server with the pstn lines |
| 19:36.52 | rue_mohr | <PROTECTED> |
| 19:36.53 | Blackthorn | When I pulled the latest cvs i pulled the * addons. and one is a graphical gui for *. Is it fairly decent? does it run from web browser, or startx? |
| 19:36.58 | bjohnson | rue_mohr: voip needs constant good connection |
| 19:37.15 | AgiNamu | Juxt, welcome to Asterisk :) |
| 19:37.18 | eKo1 | Blackthorn: no, don't care |
| 19:37.22 | AgiNamu | look in chan_iax2.c |
| 19:37.33 | bjohnson | rue_mohr: try monitoring the connection .. could be tough to pinpoint |
| 19:37.53 | Borgon | AgiNamu: last question, am reading that with asterisk as pbx it spoofs cpn, and not the ani right? |
| 19:38.12 | AgiNamu | Borgon, it depends on the CPE. |
| 19:38.14 | ast_freak | I've got my mailboxes under the default context, and I've got a default context in my dialplan with extensions for o, a, and *. The o (operator extension works, but the others don't. Can anyone give me a hand please? |
| 19:38.22 | AgiNamu | and the ILEC |
| 19:38.31 | AgiNamu | and your channel drivers |
| 19:38.33 | bjohnson | rue_mohr: everyone is dropped at the same time only once today? or a repeated thing |
| 19:38.36 | AgiNamu | look at chan_local.c |
| 19:38.51 | Borgon | ~cpe |
| 19:38.52 | jbot | cpe is, like, Customer Premises Equipment. Telephone devices such as handsets and PBXs located at the customer.s site that interface with the public network. It includes equipment such as modems, terminals and routers supplied by the telephone company, installed at customer sites and connected to the telephone network. |
| 19:38.52 | Borgon | ~ilec |
| 19:38.53 | jbot | hmm... ilec is Typically the carrier that was granted the right to provide service as a result of the breakup of AT&T. These providers are also referred to as RBOCs (Regional Bell Operating Companies) or Baby Bells. |
| 19:39.10 | Borgon | AgiNamu: am going to run to walmart and buy a headset.. test it out with free world dialup and asterisk windows version |
| 19:39.18 | Juxt | AgiNam |
| 19:39.23 | Juxt | so dtmfmode=outband |
| 19:39.24 | Juxt | ? |
| 19:40.05 | AgiNamu | For IAX, there is no setting. |
| 19:40.29 | AgiNamu | it's always out of band |
| 19:41.27 | Juxt | well why isn't it working then? |
| 19:41.36 | AgiNamu | explain your full setup |
| 19:41.51 | Juxt | 60 seats with firefly connected to a local asterisk box |
| 19:42.03 | Blackthorn | The head of the local 911 office has conacted me about the voip services I am providing. and he is wanting to do some tests. One of the things he says he gets ANI service from ilec. is there a way to set up * for ANI to 911 office? |
| 19:42.11 | Juxt | this box peers with my asterisk concentrator in a colo |
| 19:42.20 | Juxt | default codec is ilbc |
| 19:42.42 | Juxt | iax2 is used to connect firefly to local asterisk and for trunking |
| 19:42.50 | rue_mohr | VERBOSE[180236]: -- B-channel 0/2 successfully restarted on span 1 |
| 19:42.56 | AgiNamu | run iax2 debug |
| 19:43.03 | AgiNamu | and see if you see any DTMF messages |
| 19:43.04 | rue_mohr | bjohnson there is a block of those... normal? |
| 19:43.15 | ramtha | rue_mohr: yes |
| 19:43.34 | DrWho17 | Blackthorn: it should just send it over the trunks |
| 19:43.55 | AgiNamu | Juxt, how do you hit the PSTN? |
| 19:44.02 | DrWho17 | provided you have callerid setup correctly, and you are using trunks that send that info |
| 19:44.10 | AgiNamu | I'd goto the "concentrator" and do iax2 debug |
| 19:44.21 | AgiNamu | look for Rx'd DTMF packets. if you dont see anything, there's you problem. |
| 19:44.33 | AgiNamu | work back up. If you do see them, then check your outgoing |
| 19:44.39 | Blackthorn | drwho17: OK i have two out going pri's. So if the voip caller hits 911. you think it just send it out the trunk to the 911 office on a telephone number that they provide? |
| 19:45.05 | ramtha | hmm i have hidecallerid section in the user context of my sip.conf. in debian version of asterisk it works, in cvs HEAD version not |
| 19:45.09 | ramtha | what has changed there? |
| 19:45.36 | Blackthorn | drwho17: callder id does seem to work just fine incoming and outgoing. Though it reports the number the caller is from but no naming information. |
| 19:45.52 | ramtha | i meen restrictid |
| 19:47.02 | AgiNamu | to get naming information, your provide will have to do a db lookup |
| 19:47.19 | DrWho17 | Blackthorn: you need to be connected to the 911 center through dedicated trunks |
| 19:47.27 | DrWho17 | on their tandem so to speak |
| 19:47.54 | DrWho17 | also the ANI information they receive's info has to be in the SS7 database lookup |
| 19:48.27 | DrWho17 | CNAM for Caller Name, LIDB I think for location information and such (not sure, not a real SS7 Expert) |
| 19:48.30 | Blackthorn | umm. so does that mean i would have to run a dedicated line from my * box to there 911 center? |
| 19:48.38 | *** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com) |
| 19:48.43 | DrWho17 | Blackthorn: yes, that's how it normally works |
| 19:48.52 | DrWho17 | that is why E911 is such a pain |
| 19:49.03 | DrWho17 | check out www.911voip.org |
| 19:49.15 | Blackthorn | would something like a 56k dds line work... ok checking the website |
| 19:49.32 | DrWho17 | Blackthorn: I think it varies per locality |
| 19:49.33 | Juxt | ok my local asterisk is sending dtfm |
| 19:49.39 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 19:49.43 | Juxt | but the concentrator doesn't seem to show any dtfm in the log |
| 19:49.53 | DrWho17 | Blackthorn: you need to be able to update the SS7 Databases with your users info as well |
| 19:50.04 | Juxt | it shows ACK and POKE,PONG |
| 19:50.07 | AgiNamu | So on your local Asterisk, you see tx for DTMF |
| 19:50.09 | DrWho17 | we decided just not to offer it as a product offering for now |
| 19:50.15 | AgiNamu | and on the far end you dont see rx dtmf? |
| 19:50.22 | Juxt | aginamu: correct |
| 19:50.24 | AgiNamu | only ack poke and pong??? then no calls are going thru. |
| 19:50.27 | DrWho17 | interconnects to all the areas we server would be quite a big expense |
| 19:50.31 | AgiNamu | you should see a NEW and so on |
| 19:50.44 | AgiNamu | DrWho17, OM2, Intrado? :) |
| 19:51.08 | Juxt | agi: the call goes thru |
| 19:51.12 | Juxt | dtmf isn't working |
| 19:51.21 | AgiNamu | not to that machine, if you aren't getting an IAX_COMMAND_NEW :P |
| 19:52.00 | Juxt | ok now it works, weird |
| 19:54.02 | AgiNamu | well, im just saying, if you are looking at IAX traffic, and you dont see any NEW, then no new calls came in :\ |
| 19:56.00 | DrWho17 | AgiNamu: oh, yea it's worse yet |
| 19:57.19 | *** join/#asterisk afrosheen (~afro@txprotoa2.august.net) |
| 19:57.24 | afrosheen | yo |
| 20:00.30 | afrosheen | any news on the meetme delay bug? |
| 20:00.51 | ramtha | no know alese has the problem that restrictid=yes has no effect? |
| 20:01.02 | ramtha | no one... |
| 20:01.10 | ramtha | else |
| 20:01.31 | ramtha | to late for good english |
| 20:01.46 | Micc | where can I get a windows binary of iaxclient? |
| 20:01.50 | Blackthorn | thanks for the help and explanations today :) |
| 20:02.46 | afrosheen | it's never too late for gud inglish |
| 20:03.05 | ramtha | :) |
| 20:03.21 | Wazb | is there any way to change Codec for H323 |
| 20:03.34 | *** join/#asterisk Hmmhesays (negative3k@66.173.103.108) |
| 20:03.46 | *** join/#asterisk retentiveboy (~pdugas@adsl-158-43-184.asm.bellsouth.net) |
| 20:03.46 | ramtha | hmm perhaps restrictid is not working in db realtime mode... |
| 20:04.35 | *** join/#asterisk unknown1 (~unknown@ool-44c1ef43.dyn.optonline.net) |
| 20:04.58 | Juxt | wow trunk=yes makes a lot of a difference |
| 20:05.01 | unknown1 | anybody know a good and affordable supplier of cisco and polycom phones that ships same-day? |
| 20:07.31 | afrosheen | unknown1: there are alot of good polycom suppliers, voipsupply.com is one of them. |
| 20:08.11 | *** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net) |
| 20:08.30 | Wazb | is there any way to change Codec for H323 |
| 20:08.33 | unknown1 | thanks, know of any others? |
| 20:08.39 | Juxt | try froogle |
| 20:08.56 | afrosheen | unknown1: that's the best one actually, there's some yahoo store that sells polycom ip500's for around $167 each somewhere |
| 20:09.46 | Juxt | polycom phones have those horrible rubbery buttons aargh! |
| 20:10.03 | afrosheen | Juxt: and the best speakerphone in the business |
| 20:10.40 | Juxt | that is true |
| 20:10.43 | *** join/#asterisk [Outcast] (~knoppix@c-24-218-94-11.hsd1.ma.comcast.net) |
| 20:10.43 | Juxt | but buttons suck |
| 20:10.49 | *** join/#asterisk juanjoc (~juanjoc@200.73.189.82) |
| 20:11.08 | [Outcast] | has anyone worked with the Mediatrix 1204 sip gateway? |
| 20:11.16 | afrosheen | Juxt: you're not talking about the soundpoint ip, I think you're thinking of the conference phones |
| 20:12.38 | Juxt | hmm unless i saw a weird one i remember the ip 500 having rubbery buttons too |
| 20:14.30 | afrosheen | for me, the buttons are more 'plasticky', they don't click and clack like some other phones |
| 20:14.43 | afrosheen | and they require a little extra effort :) |
| 20:15.10 | afrosheen | has anyone deployed the snom 220 with the secretary keypad attachment? |
| 20:15.57 | *** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.res.rr.com) |
| 20:16.07 | Juxt | hmm i take it back, just spoke to my friend at polycom |
| 20:17.04 | afrosheen | and he said 'you take that back!' |
| 20:17.22 | Juxt | he confirmed the buttons are plasic |
| 20:17.57 | afrosheen | yeah..tps reports |
| 20:18.34 | Juxt | true |
| 20:18.45 | *** join/#asterisk lbow (~steve@wbs-146-129-17.telkomadsl.co.za) |
| 20:19.42 | *** join/#asterisk adjacent (scott@nc-65-40-81-71.sta.sprint-hsd.net) |
| 20:19.49 | afrosheen | man wtf happened to this channel, it used to be so crazy in here |
| 20:20.04 | afrosheen | now it's like a retirement community and bkw is in a rocking chair in the corner |
| 20:20.52 | adjacent | ihave moved all of my customers off and phased out voip. now i have a local PRI that they want to charge my many $k to disconnect. where would i look to find someone interested in leasing a PRI at a bargain |
| 20:21.16 | adjacent | voip providers without local phone number support in my area? |
| 20:21.26 | afrosheen | how do you phase out voip? you want to phase it in normally. |
| 20:21.55 | adjacent | yeh. it wasnt making money. so i phased it out. i couldnt afford to pay for the r&d |
| 20:22.14 | adjacent | stick to building wireless nets. which we are better at ;) |
| 20:22.27 | afrosheen | focus on the core if that's making you money, good plan |
| 20:22.49 | afrosheen | well for reselling that pri, look for providers in your area, if there are any. |
| 20:23.21 | adjacent | but i have an $800+/mo PRI that is under contract for another year. i would dump it to somone at less than my cost just to keep from getting early termination fees |
| 20:23.50 | *** join/#asterisk fugitivo (~ajf@201.255.101.68) |
| 20:24.34 | afrosheen | so you're paying $800 a month for 23 lines? |
| 20:24.40 | afrosheen | talk about a loss leader |
| 20:24.47 | tzanger | whoa |
| 20:24.47 | tzanger | __alloc_pages: 0-order allocation failed (gfp=0x1f0/0) |
| 20:24.51 | tzanger | what the hell's a 0-order allocation |
| 20:25.27 | adjacent | afrosheen: i think there are other charges on that bill. i could be off |
| 20:25.50 | afrosheen | tzanger: looks like you may be out of swap space or were when that error hit |
| 20:26.00 | tzanger | afrosheen: odd |
| 20:26.11 | afrosheen | is swap mounted? |
| 20:26.30 | afrosheen | adjacent: either way that's crazy expensive for a pri, we pay around $750 for a bonded t1 |
| 20:26.30 | tzanger | yeah, I'm only 100M into 1G of swap |
| 20:26.40 | afrosheen | tzanger: memtest86 then :( |
| 20:26.46 | adjacent | afrosheen: where are you? |
| 20:26.52 | tzanger | afrosheen: yeah ... |
| 20:26.56 | Wazb | is there any way to change Codec for H323 |
| 20:27.08 | afrosheen | adjacent: near dallas |
| 20:27.11 | *** join/#asterisk cjk_ (~cjk@80.92.75.232) |
| 20:27.21 | afrosheen | tzanger: I'll dig into it some more, may not be swap related but seems to be |
| 20:27.25 | adjacent | k. SC here. near Hilton Head. b/w isnt cheap |
| 20:27.30 | L|NUX | can some one help me |
| 20:27.37 | L|NUX | i am getting this on * |
| 20:27.38 | L|NUX | Apr 21 15:20:26 NOTICE[23626]: chan_sip.c:9293 sip_poke_noanswer: Peer 'r00t' is now UNREACHABLE! Last qualify: 1259 |
| 20:27.38 | L|NUX | Apr 21 15:21:03 NOTICE[23626]: chan_sip.c:9293 sip_poke_noanswer: Peer 'r00t' is now UNREACHABLE! Last qualify: 1466 |
| 20:27.38 | L|NUX | Apr 21 15:21:55 NOTICE[23626]: chan_sip.c:9293 sip_poke_noanswer: Peer 'r00t' is now UNREACHABLE! Last qualify: 1584 |
| 20:27.38 | L|NUX | Apr 21 15:22:33 NOTICE[23626]: chan_sip.c:9293 sip_poke_noanswer: Peer 'r00t' is now UNREACHABLE! Last qualify: 1820 |
| 20:27.39 | tzanger | afrosheen: wow thank you |
| 20:27.45 | L|NUX | what to do now :( |
| 20:27.56 | tzanger | L|NUX: don't phone as root. :-) |
| 20:28.08 | L|NUX | well its r00t :) |
| 20:28.13 | afrosheen | tzanger: http://lists.digium.com/pipermail/asterisk-users/2005-January/082827.html |
| 20:28.17 | L|NUX | its just sip account :) |
| 20:28.18 | afrosheen | looks like you're not alone |
| 20:28.46 | afrosheen | adjacent: there's dark fiber all over the place here |
| 20:29.15 | adjacent | ahh. that explains alot, then =) i need to find a dark line out of here! |
| 20:29.51 | *** join/#asterisk bamafan (~noname@fw1.ci.birmingham.al.us) |
| 20:30.08 | L|NUX | any one help me |
| 20:30.33 | afrosheen | L|NUX: is this on your local * server or a remote one |
| 20:30.59 | L|NUX | afrosheen : its on us server |
| 20:31.12 | L|NUX | dedicated server @ staminus.net |
| 20:31.40 | bamafan | Is there any way to control the number of rings on a X100P before it answers? I'd like to have the opportunity to answer the phone before the X100P card picks up. |
| 20:32.03 | Nugget | roll tide. |
| 20:32.09 | bamafan | oh yah! :) |
| 20:32.25 | *** join/#asterisk docelm0 (~docelm0@67.106.194.90.ptr.us.xo.net) |
| 20:32.45 | L|NUX | any idea :( |
| 20:33.16 | docelm0 | Whats new? |
| 20:34.18 | bamafan | You folks are my last hope. <g> |
| 20:34.26 | docelm0 | For what? |
| 20:34.51 | bamafan | Trying to determine how to control number of rings before pickup for a X100P fxo card. |
| 20:36.24 | retentiveboy | banafan: could add wait() at start of context but how would * tell that you'd answered it already, hmm... |
| 20:36.39 | malcolmd | bamafan: War Eagle! |
| 20:36.49 | bamafan | malcolmd: bah! |
| 20:37.08 | docelm0 | Im thinking the wait.. Other wise your gonna send answer.. Why why wait? |
| 20:37.10 | bamafan | retentiveboy: Surely there is a way to do this. |
| 20:37.36 | Nugget | you could buy one of those $4 "fax/modem line lockout" deals at radio shack. |
| 20:38.02 | Nugget | that way the line to the x100p would get cut off if you picked up another phone |
| 20:38.37 | *** join/#asterisk km- (pgrace@brdgw1.rttx.com) |
| 20:38.47 | km- | howdy! |
| 20:38.50 | Nugget | that, coupled with a wait in the dialplan, might do it |
| 20:38.50 | bamafan | Nugget: That sounds like an option. Can you specify some sort of lockout time for that device? |
| 20:39.03 | Nugget | no, it's just a hardware thing. if a phone is lifted, the line is cut. |
| 20:39.11 | km- | Anyone here come up with any novel solutions to the dreaded # transfer? |
| 20:39.18 | Nugget | usually you put it the other way around, so if the modem or fax is engaged the normal phones are disabled |
| 20:39.26 | km- | Users complain that they cant check voicemail on systems that use # |
| 20:39.28 | bjohnson | bamafan: a suitably long wait would give you a chance to answer and the stopping of the ringing would be all that * needs |
| 20:39.31 | Nugget | but in your case you could use it to have the "real" phone override asterisk |
| 20:39.37 | UBiQUiTY | i need my AGI script to send a DTMF ... anybody know how? |
| 20:39.40 | km- | I'm thinking of changing it to *# or #* or something like that. |
| 20:40.01 | bamafan | Nugget: The fxo card answers right after callerid is received. Isn't the "wait" entered after the call has been picked up? |
| 20:40.07 | Nugget | no |
| 20:40.14 | bjohnson | km-: google for a change to ## |
| 20:40.29 | Nugget | asterisk doesn't answer until you tell it to, or until you do something that requires answering |
| 20:40.36 | km- | bjohnson: awesome! |
| 20:41.21 | bamafan | Nugget: This'll get me started. You da man. I appreciate the advice. |
| 20:41.27 | Nugget | roll tide :) |
| 20:41.30 | bamafan | :) |
| 20:41.46 | bjohnson | km-: might not be what you want .. I remember reading about it for lines dealing with credit card machines |
| 20:41.51 | bamafan | Well kiss my grits. |
| 20:41.58 | docelm0 | UBI What are you coding in? |
| 20:42.01 | bamafan | You live in Bham proper? |
| 20:42.03 | bjohnson | I hope they're clean |
| 20:42.05 | km- | BRIAN. |
| 20:42.19 | Nugget | oneonta (way the hell north), center point, and alabaster. |
| 20:42.23 | UBiQUiTY | docelm0: im coding in php |
| 20:42.23 | km- | bjohnson: yeah, apparently bkw made a patch for it but took it down |
| 20:42.26 | Nugget | now I'm in austin texas. |
| 20:42.32 | docelm0 | What do you want to know? |
| 20:42.41 | docelm0 | Are you using PHPAGI by chance? |
| 20:42.42 | bamafan | Nugget: Is work moving you around? |
| 20:42.59 | Nugget | I moved for a job, yeah |
| 20:43.06 | UBiQUiTY | docelm0: i dont think i am |
| 20:43.11 | Nugget | I'm dug in now, though. austin is great. |
| 20:43.15 | Nugget | never leaving. :) |
| 20:43.26 | bamafan | Nugget: I've heard it's nice over there. |
| 20:43.26 | UBiQUiTY | docelm0: my php agi scripts are working, but i am unable to use the SendDTMF command |
| 20:43.49 | bamafan | Nugget: I guess I'll stay here. Born and raised here. |
| 20:43.53 | UBiQUiTY | docelm0: EXEC SendDTMF # doesnt seem to be doing anything at all |
| 20:44.46 | bamafan | Nugget: Thanks again for the help. I may be back to bend your ear again. |
| 20:45.30 | km- | why is he always friggin missing when I need him hehe |
| 20:45.52 | Strom_C | hello everyone |
| 20:47.35 | bjohnson | km-: bribe him .. rumour has it he likes chocolate (or was that ManxPower?) |
| 20:48.21 | Hmmhesays | hmmm I want to fork calls between regional locations, should I bother trying to set up SER, or should I just do it with asterisk |
| 20:48.28 | retentiveboy | anybody here using voicetronix cards? |
| 20:48.34 | Hmmhesays | sip calls that is |
| 20:49.41 | Juxt | Hmmhesay: dundi! |
| 20:49.42 | *** join/#asterisk bah (048830696@AC9951CA.ipt.aol.com) |
| 20:49.56 | Hmmhesays | Yeah, I was thinking about that |
| 20:50.18 | Juxt | i am thinking that dundi also might be a good way to make asterisk redundant |
| 20:50.23 | Hmmhesays | there will probably 100 sip calls at each regional location at any given time, and nat |
| 20:50.57 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 20:51.53 | heison | ~seen moc[toronto] |
| 20:51.54 | jbot | moc[toronto] <~mochouina@209.47.87.2> was last seen on IRC in channel #asterisk, 15h 49m 22s ago, saying: 'Hail'. |
| 20:52.59 | Hmmhesays | I hate NAT too |
| 20:53.05 | Hmmhesays | nat needs to farking die |
| 20:53.26 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 20:54.29 | *** join/#asterisk ToyKeeper (spanky@c-24-9-113-171.hsd1.co.comcast.net) |
| 20:55.35 | Bile_One | bjohnson, you have any knowledge on call files? |
| 20:56.12 | L|NUX | is there any ITSP which IAX support in Pakistan ? |
| 20:57.15 | *** join/#asterisk MatsK (~matsk@107.80-202-57.nextgentel.com) |
| 20:57.34 | *** join/#asterisk DrJolo (~chatzilla@217.153.194.10) |
| 20:58.07 | L|NUX | is there any ITSP which support IAX termination ? |
| 20:58.32 | cypromis | L|NUX: contact wasim@convergence.com.pk |
| 20:58.36 | cypromis | he will get you something |
| 20:58.58 | L|NUX | cypromis: i need US or UK based :) |
| 20:59.17 | cypromis | 22:56 < L|NUX> is there any ITSP which IAX support in Pakistan ? |
| 20:59.26 | *** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk) |
| 20:59.29 | L|NUX | sorry for that :) |
| 20:59.36 | L|NUX | what you think about voipjet.com ? |
| 21:00.08 | *** part/#asterisk oej (~oej@206.129.72.6) |
| 21:03.28 | cypromis | dunno |
| 21:03.35 | cypromis | I don't use them so can't comment :) |
| 21:03.42 | Elshar | I use voipjet, and it seems to be alright |
| 21:04.33 | *** join/#asterisk biffhero (~rob@adsl-64-172-180-238.dsl.snfc21.pacbell.net) |
| 21:04.58 | Wazb | is there any way to change Codec for H323 |
| 21:05.14 | Wazb | please help |
| 21:06.41 | eKo1 | don't use h.323 so... |
| 21:06.56 | *** part/#asterisk biffhero (~rob@adsl-64-172-180-238.dsl.snfc21.pacbell.net) |
| 21:08.53 | ariel_ | Wazb, I don't use h323 but if it's like all the other config files that asterisk uses. like sip.conf you do disallow=all then allow=ulaw or what codec you want. |
| 21:10.34 | km- | ~seen bkw_ |
| 21:10.35 | jbot | bkw_ is currently on #asterisk (7h 33m 47s) |
| 21:10.43 | ariel_ | Wazb, see this http://www.voip-info.org/wiki-Asterisk+config+h323.conf |
| 21:10.45 | km- | bkw_: wake up dude! |
| 21:11.15 | *** join/#asterisk Micc (~mic@c-24-18-35-120.hsd1.wa.comcast.net) |
| 21:12.43 | Hmmhesays | is there any good way to fork a call from a nat device with asterisk? |
| 21:12.51 | Hmmhesays | * a sip device behind NAT i should say, lol |
| 21:13.03 | *** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net) |
| 21:13.24 | ariel_ | Hmmhesays, what do you mean fork a call? |
| 21:14.32 | Hmmhesays | more or less send a reinvite to tell the endpoint to iniated the call with a second asterisk box |
| 21:14.33 | Micc | I remember when fork just meant an eating utensil. Then it was fork(), now it has some other meanings. |
| 21:14.46 | Hmmhesays | so we don't have to proxy the rtp |
| 21:15.15 | *** join/#asterisk FuriousGeorge (~brian@ool-43516ebb.dyn.optonline.net) |
| 21:15.25 | Micc | fork just doesn't seem like the most appropriate word for that. |
| 21:15.31 | harryvv | anyone have experiance with sip translastion and ser? |
| 21:15.36 | Hmmhesays | I couldnt' think of a more appropriate word |
| 21:15.39 | eKo1 | fork is also a verb Micc |
| 21:15.59 | eKo1 | Although I don't see how a call would fork. |
| 21:16.05 | Micc | eKol, right. |
| 21:16.16 | Hmmhesays | you know what i'm talking about though, did I get my point across? |
| 21:16.26 | ariel_ | Hmmhesays, I don't think asterisk has a way to do that. |
| 21:16.27 | eKo1 | I mean, 'fork a call' implies that it takes more than one direction. |
| 21:16.29 | Micc | Hmmhesays, thats what people are calling it now days. |
| 21:16.30 | Nugget | not really. |
| 21:16.37 | Nugget | I have no idea what you intend to mean by 'fork a call' |
| 21:16.56 | Hmmhesays | Yeah but then I explained what I was trying to accomplish |
| 21:17.03 | harryvv | ive heard that term but also dont exactly know what it means. |
| 21:17.04 | Qwell | Nugget: stab it repeatedly...with a fork |
| 21:17.24 | eKo1 | no no, use a spork |
| 21:17.33 | Qwell | no, that would be sporking a call |
| 21:17.54 | Qwell | which makes less sense...because how are you going to spoon a call? |
| 21:18.04 | Hmmhesays | send a reinvite to a device behind nat so you don't have to proxy the call to a second asterisk box |
| 21:18.35 | Hmmhesays | or do something to accomplish what that would |
| 21:18.42 | eKo1 | Hmmhesays: well, the problem is, reinvites don't work well behind nat...i thinkç |
| 21:19.07 | Hmmhesays | yeah and as far as I know, asterisk won't send a reinvite if you have nat=yes anyway |
| 21:19.07 | eKo1 | That's why a nat=yes is usually followed by a canreinvite=no. |
| 21:19.27 | Hmmhesays | according to a wiki, if you have nat=yes asterisk will not send a reinvite |
| 21:19.32 | *** join/#asterisk lbow (~steve@wbs-146-129-17.telkomadsl.co.za) |
| 21:20.50 | *** join/#asterisk tessier_ (~treed@210.245.97.97) |
| 21:21.01 | eKo1 | really? hmm...I'll have to check chan_sip.c to confirm that. |
| 21:21.38 | *** join/#asterisk Gh0sty (~Ghosty@81.11.210.231) |
| 21:25.09 | km- | bjohnson: hey, do you remember any more details about the ## idea? lists.digium.com, bugs.digium.com and google all reveal people talking about it, but no code actually being referenced |
| 21:25.12 | *** join/#asterisk dishwasha (~chatzilla@208.251.32.70) |
| 21:25.15 | dishwasha | Howdy |
| 21:25.19 | lbow | aaarrggh: is there ANY way to get Nufone when you have an operational issue. they are toast when trying to call South Africa (try 01127216572770 if you don't believe me). Surely I'm doing them a favour to report their issue.... |
| 21:25.19 | km- | howdy dish |
| 21:25.52 | lbow | its been like this for hours and hours |
| 21:25.58 | Strom_C | or maybe circuits to Cape Town are just down right now :) |
| 21:26.05 | harryvv | lbowl, ask jerjer he is the onwer and is not her at the moment. |
| 21:26.40 | dishwasha | anybody know why I'm getting a Proxy Authentication Required in my SIP debugs? |
| 21:27.12 | lbow | you dial a ct number, just hear garbled echo of yourself |
| 21:27.37 | dishwasha | I saw on some forums that proxy authentication is supported in the unstable CVS which I'm compiling right now, but nothing on how to enable it |
| 21:29.37 | *** part/#asterisk loick (~loick@APuteaux-151-1-47-42.w82-124.abo.wanadoo.fr) |
| 21:31.34 | km- | it just cant be that simple.... |
| 21:32.02 | dishwasha | Well, besides that point, if I have a single SIP trunk/peer defined and a single SIP client/friend defined with extension, and I have x-lite authenticate to asterisk, what does asterisk do when I make an outbound call? Does it just pass the local authentication to the actual SIP proxy (my SIPISP)? |
| 21:32.11 | *** part/#asterisk Juxt (~Juxt@64.135.20.202) |
| 21:33.40 | *** part/#asterisk bamafan (~noname@fw1.ci.birmingham.al.us) |
| 21:34.54 | *** join/#asterisk Exstatica (Exstatica@jumping.on.the.bed.are.not.umpteenmonkeys.com) |
| 21:35.07 | *** join/#asterisk Rith (~Rith@35-28-142-66.speedexpress.net) |
| 21:39.14 | *** join/#asterisk princeofdarkness (~danalien@danalien.user) |
| 21:39.20 | Hmmhesays | eKo1: would be interesting to know |
| 21:39.31 | princeofdarkness | cl |
| 21:40.17 | harryvv | dishwasha are you making two way sip calls with xlite and what router are you using? |
| 21:40.37 | Exstatica | i have voicemail setup and i put in for the mailbox the phone number and for the password i put 1234 i'm using voicemail realtime... but the it keeps saying password invalid |
| 21:40.45 | dishwasha | I'm just trying to get a good feel of how asterisk hands off the SIP to the outbound SIP line |
| 21:44.30 | harryvv | dishwasha: have you made any calls outside of the router with xlite |
| 21:44.55 | dishwasha | yes, if I proxy directly with xlite it works, if I go through Asterisk I get this other problem |
| 21:45.29 | L|NUX | i am getting this on my remote * Apr 21 16:41:03 NOTICE[23741]: chan_sip.c:9293 sip_poke_noanswer: Peer 'r00t' is now UNREACHABLE! Last qualify: 2893 |
| 21:45.33 | L|NUX | what to do now :( |
| 21:49.45 | *** join/#asterisk jcwunder (~chatzilla@b1.lrz.vpn.lrz-muenchen.de) |
| 21:49.57 | Hmmhesays | Get a better connection, or turn qualify off |
| 21:50.16 | jcwunder | bridge ISDN over ethernet ...is TDMoE the solution ? |
| 21:50.27 | ariel_ | L|NUX, yes turn off qualify=yes to no and get a better internet connection. |
| 21:51.11 | ariel_ | jcwunder, TDMoE will only work if your on the same network. |
| 21:51.25 | *** part/#asterisk lbow (~steve@wbs-146-129-17.telkomadsl.co.za) |
| 21:51.29 | L|NUX | hmm |
| 21:51.30 | L|NUX | wait |
| 21:53.13 | *** join/#asterisk Cherebrum (3NiEfYuq7c@216.32.77.10) |
| 21:53.14 | pacific | show queues |
| 21:53.24 | L|NUX | ariel_ : but it does not fix the lag |
| 21:53.44 | afrosheen | the lag is your isp's fault or your hosts' fault |
| 21:53.56 | ariel_ | L|NUX, no your right it just does not give you the notice any more. Problem is still there. |
| 21:53.59 | Cherebrum | Can anyone orriginate an international call to me in the US to my US toll free number? I need to receive an international call for my interoperability testing with my carrier. |
| 21:54.23 | L|NUX | afrosheen : hmm |
| 21:54.23 | *** join/#asterisk Legend (~Legend@24.244.142.134) |
| 21:54.45 | Strom_C | Cherebrum, I don't believe you can call US toll-free numbers from outside the US |
| 21:55.03 | Strom_C | (with the exception of countries within the NANP) |
| 21:55.09 | L|NUX | afrosheen : i have ping time 100ms |
| 21:55.14 | Cherebrum | Storm: you can. I received a call from Portugal but the ANI was from a US number |
| 21:55.17 | Borgon | Is it possible to run asterisk with a softphone on a remote pc.. and from my pc be able to talk etc? |
| 21:55.39 | afrosheen | L|NUX: 100ms isn't much lag then |
| 21:55.50 | L|NUX | afrosheen : but its doing :( |
| 21:56.15 | afrosheen | L|NUX: how are you determining lag, by your sip peer time in *'s database? |
| 21:56.51 | L|NUX | well i just watching it using cli |
| 21:57.30 | ariel_ | L|NUX, are the calls going through? |
| 21:57.32 | *** join/#asterisk telephoneman (~mike@64.207.35.66) |
| 21:57.35 | afrosheen | L|NUX: are there quality problems related to your ping or are you just thinking it will cause trouble? |
| 21:57.43 | L|NUX | well from pakistan yeh |
| 21:57.47 | L|NUX | but from my US friend nah |
| 21:58.05 | L|NUX | quality problem :) |
| 21:58.17 | dishwasha | anybody know why I'm getting SIP/2.0 407 Proxy Authentication Required? |
| 21:58.40 | L|NUX | you need to wait for authentication :) |
| 21:58.48 | afrosheen | L|NUX: we had to setup a special config in our polycom phones for 'longhaul' i.e. a connection from here in the US to our phone in korea |
| 21:59.09 | afrosheen | L|NUX: I suspect your american friend, if he's registering with your asterisk server, may need to do something similar |
| 21:59.13 | *** join/#asterisk gpai (~gpai@209-6-134-215.c3-0.lex-ubr3.sbo-lex.ma.cable.rcn.com) |
| 21:59.32 | L|NUX | hmm |
| 21:59.32 | *** join/#asterisk nine76 (~t00r@cpe-69-135-184-24.woh.res.rr.com) |
| 21:59.45 | dishwasha | What should my sip URI be? should that contain my username and password? |
| 21:59.51 | L|NUX | nine76 : ask afrosheen |
| 22:00.06 | jabbzy | hey all |
| 22:00.12 | eKo1 | dishwasha: no |
| 22:00.18 | *** part/#asterisk Cherebrum (3NiEfYuq7c@216.32.77.10) |
| 22:00.27 | dishwasha | eKo1: so it should just be sip:realm? |
| 22:00.48 | eKo1 | dishwasha: you're getting that message because the proxy is challenging the INVITE and you are sending it bogus info. |
| 22:01.17 | harryvv | Can i control what rtp ports asterisk will use? I have setup rtp port ranges in rtp.conf but is there more then that? |
| 22:01.21 | AmaDEE0_ | When I do iax2 show peers or show registry I see 'Port' is that the sorce port or dest port (the port of the IP in the 'Host' col)? |
| 22:01.26 | telephoneman | has anyone tried to compile mpg123 on a x86_64 system? |
| 22:01.27 | eKo1 | harryvv: no |
| 22:01.32 | jabbzy | could any one give me a pointer as to how i should use monitor, i'm trying to record all calls comming into the helpdesk agents, once they are connected, but need to stop the monitor after they transfer the call out |
| 22:01.33 | ScythelX | anyone know how to solve this problem....res_odbc: Error SQLConnect=-1 errno=2002 [MySQL][ODBC 3.51 Driver]Can't connect to local MySQL server through socket '/tmp/mysql.sock - I dont have a MySQL server located on the asterisk computre its at a different machine |
| 22:01.44 | dishwasha | mind if I pm you eKo1? I'd like to send a short SIP header |
| 22:01.50 | ScythelX | but my DSN is setup correctly |
| 22:01.52 | eKo1 | dishwasha: no |
| 22:02.32 | gpai | hi I am not able to make outbond call using tdm card can anyone help if I provide my config info thanks |
| 22:02.35 | harryvv | eKo1: so even if you assign say 10000-10002 that will not work? or even 10 rtp ports? |
| 22:02.53 | eKo1 | harryvv: try it and find out for yourself. |
| 22:03.07 | eKo1 | then report back to the channel on your results. |
| 22:03.21 | L|NUX | afrosheen : but my server is located in USA California |
| 22:03.41 | harryvv | i already have but only acomplished one way communications. my soho router's firewall is only limited to like 20 ports. |
| 22:03.46 | eKo1 | ScythelX: check that dsn again. |
| 22:04.03 | nine76 | Hello afrosheen. I am the individual trying to connect x-pro to L|NUX's asterisk. |
| 22:04.31 | nine76 | Both the server and myself are located in the US. ping shows less than 100ms latency |
| 22:04.32 | emrah | anyone have an idea about this strange problem? |
| 22:04.34 | Borgon | whast the asterisk win32 website? |
| 22:04.35 | emrah | http://pastebin.ca/10048 |
| 22:04.46 | nine76 | yet in asterisk,i cannot get less than 1000ms latency |
| 22:04.47 | ScythelX | eKo1: http://pastebin.ca/10049 |
| 22:05.11 | emrah | SIP/1002@default existe |
| 22:06.09 | L|NUX | ariel_ can you help nine76 |
| 22:06.11 | L|NUX | please |
| 22:06.17 | Borgon | whast the asterisk win32 website? |
| 22:06.29 | telephoneman | is anyone running * on x86_64? |
| 22:06.33 | Qwell | ~google asterisk win32 |
| 22:07.03 | UBiQUiTY | how can i get an AGI script to run from a call file? |
| 22:07.28 | UBiQUiTY | i thought all i needed was to specify a context and an extension |
| 22:07.37 | UBiQUiTY | but it doesnt seem to be running |
| 22:08.00 | eKo1 | ScythelX: looks fine to me. Can you connect to it through another program besides *? |
| 22:08.37 | *** join/#asterisk MarlboroMan (~bob@147.202.35.253) |
| 22:08.52 | MarlboroMan | I was looking through voip-info.org, and noticed Asterisk Realtime for the first time - with this, is it possible to implement roaming extensions? |
| 22:09.06 | ScythelX | eKo1: not really sure how else to test it |
| 22:09.20 | Qwell | MarlboroMan: You don't need realtime to do "roaming extensions", assuming it is what I think it is |
| 22:09.41 | MarlboroMan | Ability for people to have extensions follow them to whatever phone they happen to be in front of. |
| 22:10.03 | Qwell | well, unless you're doing rfid tracking or something... |
| 22:10.07 | Bile_One | UBiQUiTY look on the wiki and you'll see how to execute an application using call files. |
| 22:10.18 | nine76 | had to have been network congestion. 95ms now. I hope it was congestion on my end and not the colocated servers end :-/ |
| 22:10.19 | UBiQUiTY | i read it |
| 22:10.37 | UBiQUiTY | it is making the call, but it isnt executing my agi |
| 22:10.47 | *** join/#asterisk biffhero (~rob@adsl-64-172-180-238.dsl.snfc21.pacbell.net) |
| 22:10.51 | Bile_One | UBiQUiTY, then you seen the part that starts with APPLICATION: ? |
| 22:10.54 | UBiQUiTY | its almost as if its completely ignoring the context instructions i gave |
| 22:11.09 | UBiQUiTY | hmm... maybe not... i'll go look again |
| 22:11.25 | Bile_One | UBiQUiTY have you set the file to be run under asterisk 0777? |
| 22:11.46 | Bile_One | later doods! |
| 22:11.51 | UBiQUiTY | asterisk is running as root |
| 22:11.52 | UBiQUiTY | so ya |
| 22:11.54 | UBiQUiTY | and |
| 22:12.05 | UBiQUiTY | oh well... ur leaving? thanx for the help. |
| 22:12.05 | Bile_One | asterisk should not be running as root! |
| 22:12.26 | eKo1 | ditto |
| 22:12.29 | Bile_One | fix that first. |
| 22:12.31 | UBiQUiTY | ya i dont want it to run as root, but thats how it is right now... mark spencer told me he runs it that way |
| 22:12.33 | twisted[work] | MarlboroMan, I have roaming extensions working, just w/o message waiting indication |
| 22:12.38 | UBiQUiTY | hmmmmmmmm |
| 22:12.40 | twisted[work] | MarlboroMan, all in dialplan logic ;) |
| 22:12.47 | Bile_One | later all. |
| 22:12.55 | Qwell | twisted[work]: Are you using rfid auth? :p |
| 22:13.04 | twisted[work] | Qwell, no, the user logs into the phoen |
| 22:13.05 | twisted[work] | er phone |
| 22:13.08 | Qwell | lame :P |
| 22:13.28 | Qwell | twisted[work]: I was being sarcastic earlier. |
| 22:13.35 | twisted[work] | heh |
| 22:13.35 | ScythelX | eKo1: if I.. mysql 10.0.18.1 |
| 22:13.35 | ScythelX | ERROR 2002: Can't connect to local MySQL server through socket '/tmp/mysql.sock' |
| 22:13.38 | eKo1 | UBiQUiTY: If Mark jumps of the golden gate bridge, would you also? |
| 22:13.47 | twisted[work] | user sits down, dilas *99xxxx where xxxx is their extension |
| 22:13.49 | ScythelX | i dont understand why its trying to use the socket |
| 22:13.50 | Qwell | eKo1: I would. He might know something I don't. |
| 22:13.56 | twisted[work] | they then enter their password, and their extension is moved to that phone |
| 22:13.58 | Qwell | twisted[work]: kinda what I figured |
| 22:14.04 | twisted[work] | until either they do the same thing again, or log into a different phone |
| 22:14.23 | Qwell | twisted[work]: that sends ACD and everything to them? |
| 22:14.26 | AgiNamu | hey, like how much would a DS3 connect cost? average (usa)? |
| 22:14.34 | Qwell | AgiNamu: $80k? |
| 22:14.36 | eKo1 | ScythelX: well for some reason it is trying to connect locally. |
| 22:14.37 | twisted[work] | Qwell, that extension takes on all of their stuff, with the exception of MWI |
| 22:14.40 | AgiNamu | monthly? |
| 22:14.41 | UBiQUiTY | lol... no, im not gonna jump off the bridge with mark... |
| 22:14.44 | UBiQUiTY | however |
| 22:14.47 | Qwell | AgiNamu: something like that, heh |
| 22:14.48 | UBiQUiTY | now it DOES work |
| 22:14.52 | UBiQUiTY | i dunno ... |
| 22:14.59 | Qwell | AgiNamu: dunno, I (very briefly) looked it up the other day... |
| 22:15.09 | Qwell | twisted[work]: Got your dialplan logic anywhere? |
| 22:15.16 | Qwell | and, why doesn't MWI work, out of curiousity? |
| 22:15.16 | twisted[work] | Qwell, yeah, on my server ;) |
| 22:15.25 | Qwell | twisted[work]: I mean, somewhere public. :p |
| 22:15.26 | twisted[work] | because MWI is fed from the channel driver |
| 22:15.34 | Qwell | ahh |
| 22:15.35 | eKo1 | A DS3 costs 80,000 USD per month? Dang. |
| 22:15.36 | twisted[work] | I deal strictly in dialplan logic to make it work |
| 22:15.57 | emrah | anyone has an idea for me |
| 22:15.58 | emrah | ? |
| 22:16.08 | twisted[work] | although, I *COULD* code in some variable functionality for SIP to transfer MWI when variables are set :P |
| 22:16.13 | Qwell | heh |
| 22:16.14 | AgiNamu | My friend works at a school, and they have a voice DS3, but only 50 staff members. |
| 22:16.24 | AgiNamu | and im trying to figure out wtf |
| 22:16.31 | biffhero | porn |
| 22:16.38 | twisted[work] | ie, when you set mwixxxx (mwi1001) to a channel, it would change the MWI location in the structure. |
| 22:16.39 | eKo1 | emrah: eat your enchiladas on top of your plate. |
| 22:16.42 | twisted[work] | but that's another date. |
| 22:16.42 | Qwell | twisted[work]: setvar(MWI${EXTEN}) |
| 22:16.43 | Qwell | heh |
| 22:16.44 | ScythelX | they prolly get a discount for being an educational inst. |
| 22:16.45 | twisted[work] | s/day |
| 22:16.52 | twisted[work] | bbl |
| 22:17.11 | AgiNamu | scy, yea, they pay like 15% |
| 22:17.22 | AgiNamu | even so, A DS3 is what? 674 voice channels? |
| 22:17.30 | AgiNamu | for 50 people, that's a lot. |
| 22:17.33 | eKo1 | but isn't a ds3 for 50 people overkill |
| 22:17.38 | AgiNamu | yea |
| 22:17.43 | Qwell | I bet he's lying, or wrong about what they have |
| 22:17.45 | ScythelX | it doesnt cost 80,000 a month |
| 22:17.56 | AgiNamu | he's got pictures of it. he knows his shit |
| 22:17.59 | Qwell | ScythelX: That what the first hit on google said when I looked the other day. heh |
| 22:18.18 | eKo1 | Unless the school is just a front for a phone scam business. |
| 22:18.27 | ScythelX | heh |
| 22:18.49 | AgiNamu | lol |
| 22:19.12 | dishwasha | Depending on what you're porting over it, a DS3 loop costs about $8,000/month |
| 22:19.30 | eKo1 | If it's just voice, how much? |
| 22:20.03 | dishwasha | dunno, we use DS3 for p2p T1s |
| 22:20.19 | Nugget | TLA overload. |
| 22:20.36 | biffhero | I have two sipura841 phones. They are each behind different NAT networks. Can * help them find each other? |
| 22:20.42 | dishwasha | so there's extra port and PVC costs on top of the loop |
| 22:22.49 | telephoneman | anyone running Asterisk on x86_64? |
| 22:24.02 | telephoneman | anyone running Asterisk? |
| 22:24.07 | telephoneman | anyone? |
| 22:24.14 | eKo1 | no no, we all run ser here. |
| 22:24.22 | eKo1 | isn't this #ser? |
| 22:24.33 | *** part/#asterisk roamer323 (~sing@toronto-HSE-ppp4075335.sympatico.ca) |
| 22:24.58 | eKo1 | telephoneman: why do you want to run * on x86_64? |
| 22:25.01 | dishwasha | So this isn't the shift-8 users anonymous? |
| 22:25.26 | telephoneman | why would I want to run an Opteron processor? |
| 22:25.40 | eKo1 | certaintly not for *. |
| 22:25.48 | dishwasha | because you like big butts and you cannot lie |
| 22:25.53 | eKo1 | lol |
| 22:26.08 | eKo1 | quad opteron? |
| 22:26.16 | eKo1 | that'd be hella-sweet |
| 22:26.19 | telephoneman | and why not? after trying several MB for spandsp, I found one that works! |
| 22:26.26 | jabbzy | could any one give me a pointer as to how i should use monitor, i'm trying to record all calls comming into the helpdesk agents, once they are connected, but need to stop the monitor after they transfer the call out |
| 22:26.54 | eKo1 | telephoneman: well, if you do get it working well, post your results on the wiki |
| 22:26.57 | *** join/#asterisk mog_home (~mog_home@146.229.180.196) |
| 22:27.07 | *** join/#asterisk roamer323 (~sing@toronto-HSE-ppp4075335.sympatico.ca) |
| 22:27.23 | telephoneman | it works! everything is great! i just can't compile mpg123... |
| 22:27.39 | telephoneman | it doesn't like the assembler pushl instruction |
| 22:27.40 | eKo1 | ah! so don't use it. mpg123 sucks and is discontinued. |
| 22:27.59 | telephoneman | but try installing AMP w/o mpg123... |
| 22:28.18 | eKo1 | AMP?! ugh. |
| 22:28.26 | dishwasha | Cool, now that I have Asterisk from CVS unstable, I see Proxy-Authenticate stuff; just don't know how to configure it |
| 22:28.45 | fugitivo | telephoneman: do you want me to try to compile it? i'm using Athlon 64 |
| 22:29.01 | *** part/#asterisk biffhero (~rob@adsl-64-172-180-238.dsl.snfc21.pacbell.net) |
| 22:29.02 | telephoneman | i need a configuration GUI in a multi-tenant environment. suggestions? |
| 22:29.32 | eKo1 | gvim? |
| 22:29.35 | telephoneman | fugitivo: yes, please try it |
| 22:29.45 | telephoneman | heh heh |
| 22:29.45 | fugitivo | telephoneman: i'm using gentoo |
| 22:30.01 | telephoneman | but the same gcc compiler as RH? |
| 22:30.09 | fugitivo | i don't know |
| 22:30.12 | fugitivo | i don't use RH |
| 22:30.22 | telephoneman | but the compiler is the same... |
| 22:30.39 | *** join/#asterisk MikeJ[Laptop] (~jayden@pcp02795302pcs.roylok01.mi.comcast.net) |
| 22:30.46 | R3DB0x | has anyone used switchvox? |
| 22:30.48 | fugitivo | maybe, but i'll use the mpg123 from the gentoo portage |
| 22:31.10 | telephoneman | in asterisk, i think you do a make mpg123 |
| 22:31.40 | *** join/#asterisk ngb (~ngb@200.49.156.89) |
| 22:31.42 | ngb | hello |
| 22:31.48 | telephoneman | hello |
| 22:31.54 | ngb | some one configure the motorola vt1000 ? |
| 22:32.00 | *** join/#asterisk jason^ (jason@acs-24-154-127-188.zoominternet.net) |
| 22:32.14 | telephoneman | is that the rs232 terminal? |
| 22:32.15 | eKo1 | Is that a question or a command? |
| 22:32.32 | ngb | question |
| 22:32.43 | jason^ | what is a good phone for voip with asterisk that is in a decent price range? |
| 22:32.56 | telephoneman | Polycom 300 |
| 22:33.10 | telephoneman | no mic on the speakerphone, tho |
| 22:33.11 | fugitivo | telephoneman: mpg123 0.59s-r9 compiles without problems (Athlon64 gentoo) |
| 22:33.14 | ngb | some one know the motorola vt1000 ? |
| 22:33.48 | telephoneman | fugitivo: ok, thanis |
| 22:33.52 | telephoneman | ok, thanks |
| 22:34.03 | ngb | fugitivo hablas espaņol ? |
| 22:34.51 | fugitivo | yes |
| 22:35.10 | ngb | fugitivo conoces el motorola vt1000 |
| 22:35.11 | ngb | ? |
| 22:35.16 | fugitivo | no |
| 22:35.21 | ngb | ok |
| 22:35.22 | jason^ | telephoneman: how about one with a mic? |
| 22:38.02 | jontow | est-ce que tu as une VT-100?! |
| 22:38.30 | jontow | (nevermind me.) |
| 22:42.16 | *** join/#asterisk bkw_ (~brian@bkw.developer.and.friend.of.asterisk) |
| 22:42.16 | *** mode/#asterisk [+o bkw_] by ChanServ |
| 22:43.34 | L|NUX | do voipjet.com give call recording service ? |
| 22:45.15 | rue_mohr | bjohnson thanks, I'll come back if I can find out more |
| 22:46.06 | L|NUX | can some one help me with this notice why i am getting this |
| 22:46.06 | L|NUX | Apr 21 17:37:10 NOTICE[25601]: rtp.c:541 ast_rtp_read: Unknown RTP codec 72 received |
| 22:46.06 | L|NUX | Apr 21 17:37:13 NOTICE[25601]: rtp.c:541 ast_rtp_read: Unknown RTP codec 72 received |
| 22:46.12 | L|NUX | i am using gsm codec |
| 22:51.14 | ariel_ | L|NUX, I think that if you go to the voipjet site you will see they only do ulaw and ilbc for codec. I have not been able to get gsm to work with them. |
| 22:52.27 | L|NUX | ariel_ : i am right now just doing sip to sip not using any provider though |
| 22:52.45 | L|NUX | ariel_ : but i want to know why i am getting this error |
| 22:56.04 | dishwasha | OMG this is really getting on my nerves |
| 22:57.53 | dishwasha | if I'm making an outbound SIP call, how can I get it to appear as though the SIP call is coming from a SIP line defined in asterisk rather than my SIP client? |
| 22:57.55 | *** join/#asterisk syslod (~yurplsl@65.114.15.71) |
| 22:58.37 | syslod | Hello. |
| 23:03.18 | L|NUX | dishwasha : callerid = something |
| 23:03.39 | L|NUX | in your sip.conf [exten] |
| 23:03.48 | *** join/#asterisk jetx (~jetx@adsl-64-219-216-41.dsl.hstntx.swbell.net) |
| 23:04.17 | dishwasha | hrm, I'll try that, thanks for the advice |
| 23:06.36 | Qwell | So, when you hit the mute button on an analog phone while in MeetMe, it says "Muted" or "Unmuted". Is it possible to get this functionality during all calls? |
| 23:07.07 | AmaDEE0_ | Why the different port numbers for a Host when I do iax2 show peers and iax2 show registrey |
| 23:07.08 | AmaDEE0_ | ? |
| 23:08.39 | Qwell | hmm, its not seeming to do so locally...wonder how bkw did it on his |
| 23:09.52 | Qwell | maybe I'm wrong |
| 23:18.21 | *** join/#asterisk ptg123 (~ptg123@001-759-866.area1.spcsdns.net) |
| 23:26.06 | dizzydiffi | i got it to work |
| 23:26.12 | dizzydiffi | yo yo |
| 23:27.22 | *** join/#asterisk dizzydiffi (dizzydiffi@adsl-70-240-211-145.dsl.hstntx.swbell.net) |
| 23:27.29 | dizzydiffi | sup peps |
| 23:27.36 | *** join/#asterisk TechDawg (voipnewbie@168.215.180.100) |
| 23:27.58 | dizzydiffi | has anyone done radius authenication with Asterisk |
| 23:28.21 | TechDawg | Okay, I have the FXS system working, but now I'm having issues with the FXO system. Getting several errors. |
| 23:29.30 | TechDawg | Uhm, do I have to have a sound card in each server? |
| 23:29.36 | timecop | how can I tell which codec a currently live h323 channel is using? |
| 23:36.34 | Mavvie | jbot: conference? |
| 23:36.35 | jbot | methinks conference is IAX2/asterisk@switch-1.nufone.net/4569 |
| 23:36.56 | tzanger | ? |
| 23:37.04 | tzanger | I didn't know nufone had a conf |
| 23:37.23 | Mavvie | oh, isn't that the developers conference number? |
| 23:37.31 | tzanger | no |
| 23:37.34 | Mavvie | bugger. |
| 23:37.41 | tzanger | IAX2/guest@switch-3.asterlink.com/996 |
| 23:38.14 | TechDawg | Okay, figured out that problem. |
| 23:40.57 | Mavvie | sounds pretty dead. |
| 23:41.22 | TechDawg | Apr 21 18:42:51 WARNING[381]: chan_iax2.c:5553 socket_read: Call rejected by xxx.xxx.xxx.xxx: No authority found. What did I miss in the extensions or iax setup? |
| 23:41.43 | Borgon | hello |
| 23:42.15 | Borgon | IS it possible to have asterisk installed on a remote server and then use a softphone from a different pc to make out and incoming calls? |
| 23:43.10 | dishwasha | Okay, I have a very simple question. How do I get local numbers to dial out on a SIP line? |
| 23:43.15 | dishwasha | exten => _NXXXXXX,1,Dial(SIP/line1) is what I have in extensions.conf |
| 23:43.54 | dishwasha | do I have to pass the number dialed somehow? |
| 23:44.03 | *** join/#asterisk simonides (simon@byte.unitycode.org) |
| 23:50.00 | niZon | jeez |
| 23:50.17 | niZon | iax.cc is taking their sweet time with DIDs in 204 |
| 23:50.29 | timecop | Borgon: of course |
| 23:50.35 | niZon | they've been saying sometime this week for the past 3 weeks |
| 23:50.35 | timecop | Borgon: thats the whole point? |
| 23:51.12 | timecop | dishwasha: you want ${EXTEN} |
| 23:51.15 | timecop | instead of "line1" or whatever. |
| 23:51.30 | Borgon | timecop: sorry am new to this, i just want to remain anonymous |
| 23:51.59 | dishwasha | timecop: and what does ${EXTEN} represent? I know that shows the extention, but I don't want to dial an extension |
| 23:52.01 | timecop | dishwasha: and if "line1" is a sip account, dial(sip/${exten}@line1) |
| 23:52.08 | dishwasha | I'm really confused on extensions and flows |
| 23:52.12 | timecop | dishwasha: exten is the nubmerx after nxxxxxx |
| 23:52.23 | timecop | another words its the shit you jsut dialed |
| 23:52.24 | dishwasha | ooooh, I thought it was the extension calling from |
| 23:52.26 | timecop | no |
| 23:52.27 | dishwasha | k, cool |
| 23:52.43 | timecop | Borgon: well, it works very well. |
| 23:53.29 | dishwasha | HOLY SHIT! |
| 23:53.35 | dishwasha | THANK YOU timecop |
| 23:53.37 | dishwasha | Thank you |
| 23:53.47 | *** join/#asterisk ToyMan (~konversat@user-12lcqur.cable.mindspring.com) |
| 23:53.50 | dishwasha | I made my first successful SIP call from x-lite on my workstation through asterisk to my cell phone |
| 23:54.02 | dishwasha | if you were hear I would hug you (kisses are not my forte) |
| 23:54.06 | dishwasha | here even |
| 23:54.09 | timecop | u |
| 23:54.10 | timecop | huhu |
| 23:56.43 | Borgon | timecop: so i can have a softphone on my pc.. and asterisk would be on the remote pc making all the calls right? |
| 23:56.59 | Borgon | am testing asterisk on my winxp pc.. all that is needed is an ethernet conn right? |
| 23:57.00 | TechDawg | What hardware would I need to allow * to plug into an ISN BRI PBX? |