| 00:00.26 | *** join/#asterisk jjhall (~jjhall@24-116-128-106.cpe.cableone.net) |
| 00:00.41 | brif8 | curious to what network / infrastructure does nufone, etc.. connect to, to offer the rates as low as 3.9c/min ? |
| 00:01.09 | brif8 | telco still charges for T1 calls +/- 5-7 c/min ? |
| 00:01.09 | epreston | what would case a "Dial End" event on the channel right at the beginning of "Dial" app? |
| 00:01.15 | epreston | s/case/cause/ |
| 00:01.38 | killerbee | hey is is me.. or is fwd screwy |
| 00:02.02 | *** join/#asterisk zwi (~zwi@216.88.131.43) |
| 00:02.33 | *** join/#asterisk Tili (Tili@202-133-65-43-dialup.sat.net.pk) |
| 00:02.36 | tessier | Anyone know what digiums lead time on T1 cards normally is? |
| 00:02.48 | brif8 | <PROTECTED> |
| 00:02.49 | file[desk] | two years! |
| 00:02.59 | ManxPower | Nufone is 2.9/cents/min for USA calls |
| 00:03.25 | habakuk | if you can register to them |
| 00:03.36 | ManxPower | brif8: Nufone has some LARGE corporate customers and so get VERY good wholesale rates from the telcos. |
| 00:03.42 | brif8 | even 2.9 c/min how, since a telco charges 5-7 c/min for a T1 call |
| 00:03.53 | ManxPower | brif8: They don't. |
| 00:03.54 | epreston | anyone here know the chan driver code well? I could use some help/info in a pvt chan |
| 00:04.02 | ManxPower | brif8: Where do you get those numbers from???? |
| 00:04.08 | brif8 | sprint |
| 00:04.09 | mitchel_ | does anyone know a company with a better charge per minute then nufone? |
| 00:04.27 | ManxPower | Heck, I get 7/cents/min as a RETAIL customer of IDT |
| 00:04.37 | file[desk] | mitchel_: nope. :) |
| 00:04.44 | ManxPower | And TalkAmerica had 5/cents/min for a long time. |
| 00:04.49 | machinehd | someone in here must use Asterisk for sip -> pstn calls? |
| 00:04.58 | brif8 | I'm looking to get a T1 line and sprint (our local telco) says a T1 is $795 per month plus 6c/min |
| 00:05.01 | Carp | anyone know who wrote agi-ccard.agi? |
| 00:05.10 | file[desk] | bkw_ did |
| 00:05.14 | ManxPower | brif8: Tell them you will comit to 10,000mins/month. |
| 00:05.23 | jjhall | My POTS provider charges $.05 per minute, with a $20 per month cap. You can bet they are paying less than what Nufone charges... LOL |
| 00:05.36 | ManxPower | They will give you the T-1 for free and charge you some horribly low (sometimes less than 1 cent/mon) rate. |
| 00:05.56 | ManxPower | mon == min, of course. |
| 00:06.02 | brif8 | wooah !!!! if you comit to only 10,000 mins |
| 00:06.19 | ManxPower | brif8: I believe that Nufone does more than 10,000 mins/month |
| 00:06.39 | brif8 | what happens if you don't make the full 10,000 min per month ? |
| 00:06.41 | ManxPower | brif8: Heck, we got 5/cents/min from Sprint with no usage comitments. |
| 00:06.42 | jjhall | I'm sure some of Nufone's customers use more than 10K per month.. |
| 00:06.50 | ManxPower | brif8: Then you pay some penalty. |
| 00:07.01 | mitchel_ | whats wrong with stringing 24 lines from a normal pots to asterisk? instead of getting a t1 |
| 00:07.06 | brif8 | roughlt what king of penalty ? |
| 00:07.17 | ManxPower | It will be all spelled out in lawereze in the 400 page contract they will make you sign. |
| 00:07.33 | *** join/#asterisk todd- (ewdmfw2t5n@fries.net) |
| 00:07.37 | *** join/#asterisk rd (nobody@sivka.rdy.com) |
| 00:07.39 | ManxPower | brif8: Prolly no more than just pading your bill up to 10,000mins for that month |
| 00:07.41 | *** join/#asterisk rd (nobody@sivka.rdy.com) |
| 00:07.48 | brif8 | aah!! so you'll have to sign a contract also. |
| 00:08.03 | ManxPower | brif8: Of course. That's the way you get good rates. |
| 00:08.10 | rd | anybody can help me with asterisk config? |
| 00:08.11 | citats | brif8: of course... verbal commitments mean squat |
| 00:08.18 | ManxPower | Good rates require contracts and monthly usage comitments |
| 00:08.19 | jjhall | mitchel_: Sound quality would be a lot better with T1, less echo problems, hardware is cheaper, monthly line charges would be cheaper... Lots of advantages. POTS lines really suck compared to the other options available. |
| 00:08.27 | brif8 | ok what if you commit to more than 10,000 min, would you get an even better rate ? |
| 00:08.38 | ManxPower | brif8: I'm sure you will. |
| 00:08.38 | *** join/#asterisk snewpy (~markl@dsl-202-173-156-4.vic.westnet.com.au) |
| 00:09.05 | jjhall | brif8: Obviously, the more you commit to spend with a company, the better the perks they will give you in order to make sure they get your business rather than their competator. |
| 00:09.15 | brif8 | thanks ManxPower: what about costs on a 1-800 number, same applies ? |
| 00:09.28 | citats | brif8: everything is negotiable |
| 00:09.52 | mitchel_ | jjhall: it seems to me that a t1 is actually more expensive then just a business line? |
| 00:10.00 | mitchel_ | jjhall: on a cost per line |
| 00:10.00 | brif8 | jjhall: understood, but sprint seems to be the only ones local, here, at least |
| 00:10.13 | Matthew_I | rd: it annoys people when you come in and ask to ask or ask questions like "can some one help me?" Just ask your question |
| 00:10.23 | citats | mitchel_: it depends on the carrier... around here at about 6-8 lines the t1 becomes cheaper |
| 00:10.36 | ManxPower | brif8: Generally yes. |
| 00:10.46 | jjhall | mitchel_: For one or two lines, yes. But depending on the options available, once you reach the 4-5 line point, T1 actually becomes cheaper. It takes more lines to reach that point depending you your exact location. |
| 00:10.59 | ManxPower | brif8: Large companies have different t-1s for local calling .vs. toll calling. |
| 00:11.13 | ManxPower | brif8: Where are you located? |
| 00:11.23 | brif8 | thanks guys, I'm going to give my sprint rep a BIG talking to tomorrow, as to why our rates are so bad, |
| 00:11.29 | brif8 | ManxPower: Citrus County, FL |
| 00:11.47 | ManxPower | brif8: Most LD carriers will service most points in the USA. |
| 00:11.54 | jjhall | brif8: Sprint may have to give you the T1 loop themselves, but who you connect your T1 to, and who you get service from, can usually be different. |
| 00:12.30 | ManxPower | brif8: Get quotes from 3-4 carriers then play them off against each other. "Well AT&T will give me toll calling for 1 cent/min less than you are. Whatcha gonna do about it, sales boy?" |
| 00:12.31 | brif8 | jjhall: Oh! ok then if I may ask who and what are you all paying, for T1s ? |
| 00:13.10 | ManxPower | brif8: The LD company will normally pay for the T-1 if you comit to even a reasonably small per month comitment. |
| 00:13.15 | jjhall | brif8: In my case here, Qwest provides the copper loops for the T1, and Sprint, MCI, TouchAmerica, and numerous others can provide the service. |
| 00:13.23 | *** part/#asterisk rgagnon_ (~rgagnon@rrcs-sw-24-173-78-2.biz.rr.com) |
| 00:13.31 | *** join/#asterisk Kumbang (~unknown@167.205.22.54) |
| 00:13.37 | ManxPower | jjhall: But do you pay Qwest for the T-1 or does the LD carrier do that? |
| 00:13.47 | *** part/#asterisk Matthew_I (~matthew@64-89-121-30.arpa.kmcmail.net) |
| 00:14.07 | rd | mattew: okay, here we go |
| 00:14.13 | epreston | what exactly does read_channel bridge=1 mean? |
| 00:14.16 | jjhall | ManxPower: Depends on the contracts. I don't handle that side of things for my company. |
| 00:14.19 | ManxPower | Now, IN-STATE long distance or IN-LATA long distance is a totally different thing. |
| 00:14.28 | brif8 | man I'm sure glad I asked I knew my deal was high, but now it's down right THEFT, it also makes the 2.9c more understandible |
| 00:14.47 | mitchel_ | does anyone have an idea for me? i get 1.7C's a minute from my t1 provider, but i'd prefer something lower if possible, also unlimited outgoing calls (where i just pay per minute...i.e. through some service online) |
| 00:14.54 | rd | I'm trying to run asterisk on freebsd. right now my network has only one voip phone - cisco 7940 (skinny) |
| 00:14.55 | ManxPower | brif8: Yes, but no-contract, no-usage comitment will always cause higher rates. |
| 00:15.16 | rd | I was trying to use example configs provided by asterisk distribution |
| 00:15.31 | ManxPower | mitchel_: 1.7c/min is better than most VoIP companies will provide. |
| 00:15.35 | rd | now it _kinda_ works, but when I try to dial for voicemail, asterisk dumps core |
| 00:15.53 | ManxPower | rd: Asterisk is only supported on Linux. |
| 00:16.08 | jjhall | brif8: Long and short of it, is the more you plan to use, the better your price point per minute will be, and the more perks you can get. Figure out (realistically) what you plan to use and start calling carriers and getting quotes. |
| 00:16.11 | ManxPower | There are a few *BSD users that have gotten it to work, but they are pretty much on their own. |
| 00:16.26 | ManxPower | rd: Recent CVS versions seem to run better on *BSD |
| 00:16.36 | rd | manxpower: in other words, if I decide to go with FreeBSD, I'm on my own and people here wouldn't be able to help, right? |
| 00:16.47 | ManxPower | brif8: Your local phone company will almost always have high LD rates. |
| 00:17.04 | *** join/#asterisk adelorenzo (~adelorenz@201.7.92.211) |
| 00:17.08 | ManxPower | rd: There might be one or two people that come here occasionally that run *BSD, but that's all. |
| 00:17.31 | jjhall | rd: Some people here will be able to help, but most of us are using various flavors of Linux. |
| 00:17.32 | ManxPower | brif8: Let the LD company deal with the T-1. |
| 00:17.54 | rd | manxpower: I suspect that it has nothing to do with bsd/linux. It's probably something dumb in my config files, but thanks for your help anyway |
| 00:18.05 | ManxPower | rd: Asterisk should never dump core. |
| 00:18.12 | ManxPower | rd: What verison of Asterisk are you using? |
| 00:18.23 | rd | 0.7.2 |
| 00:18.42 | ManxPower | rd: SCCP is alpha quality, so that can be the problem too. |
| 00:18.58 | rd | would it be helpfull, if I show a stack trace? |
| 00:19.11 | killerbee | hey.. is the SIP to SIP problem still in CVS? |
| 00:19.29 | ManxPower | rd better to post it to the astrisk-dev mailing list and file a bug with bugs.digium.com if you are using chan_skinny |
| 00:19.46 | ManxPower | killerbee: Supposed to be fixed today or yeterday. |
| 00:20.06 | killerbee | humm... |
| 00:20.10 | rd | manxpower: the problem is - I'm not sure it's an asterisk bug. it's probably misconfiguration on my side, since it's obvious that I'm a newby when it comes to voip |
| 00:20.17 | todd- | SCCP? |
| 00:20.22 | killerbee | ManxPower ..iax seems ok |
| 00:20.24 | ManxPower | killerbee: I don't keep the messages from the asterisk-cvs mailing list do I don't know for sure. |
| 00:20.35 | todd- | rd: are you doing CME ? |
| 00:20.44 | ManxPower | todd-: SCCP aka Skinny aka Cisco secret protocol for use with Cisco Call Manager |
| 00:20.50 | rd | todd: what's a CME? |
| 00:21.02 | ManxPower | todd-: Some Cisco phones only run SCCP/Skinny not SIP |
| 00:21.04 | todd- | cisco call manager = CME |
| 00:21.07 | todd- | right |
| 00:21.13 | todd- | I have a setup I've just turned up |
| 00:21.23 | rd | todd: ahh |
| 00:21.25 | todd- | that uses a cisco with CME to set a dial-peer as sip |
| 00:21.59 | todd- | if you have a cisco with CME you can set an option on the sip dial peer 'dtmf-relay sip-notify' and I've a patch as a wip that works to receive dtmf from the cisco sccp phones |
| 00:22.04 | rd | todd: I don't think so. do you have a minute to take a look at the config files? I believe they were created from your examples |
| 00:22.29 | todd- | my examples? I have not put any config files up .. perhaps you may be thinking of someone else |
| 00:22.34 | tclark | gota luv that ml post from Don Feuer hearing that there are lota issue with AIX providers |
| 00:22.46 | ManxPower | tclark: *nod* LOL! |
| 00:22.54 | rd | sivka# grep todd * |
| 00:22.54 | rd | extensions.conf:;;exten => s,3,Dial(IAX/jtodd@nufone/${ARG1},100,T) |
| 00:22.54 | rd | indications.conf:; 2003-04-24 05:06 GMT jtodd@loligo.com |
| 00:22.54 | rd | logger.conf:; 2003-04-24 05:06 GMT jtodd@loligo.com |
| 00:23.00 | rd | perhaps it's a different todd |
| 00:23.12 | todd- | I'm going to guess that you have a cisco call manager ? |
| 00:23.23 | bkw_ | rd jtodd is jtodd |
| 00:23.26 | bkw_ | todd- is someone else |
| 00:23.27 | rd | ahh |
| 00:23.29 | todd- | I'm `Todd Fries todd@fries.net http://FreeDaemonConsulting.com' and if you really wanna pic http://todd.fries.net |
| 00:23.31 | tclark | like they can gurantee ppl isp svcs wonder how he spells voip :) |
| 00:23.33 | rd | bkw: my bad |
| 00:23.43 | ManxPower | it is. he's jtodd here on irc |
| 00:23.52 | machinehd | what is the best gui? any work with a mysql setup? |
| 00:23.58 | todd- | and for whatever it's worth I just spent time in the last 20hrs getting asterisk to work on OpenBSD .. |
| 00:24.06 | rd | todd: honestly, last thing I care right now is somebodies picture :-) |
| 00:24.06 | bkw_ | todd- kewl |
| 00:24.17 | rd | todd: did it work? |
| 00:24.21 | blll | asterisk on openbsd would be erection-worthy |
| 00:24.40 | bkw_ | haha |
| 00:24.41 | ManxPower | todd-: Did you find much help? |
| 00:24.44 | tclark | killerbee: like after latest fix to rtp.c still see nonsense like this .. |
| 00:24.46 | todd- | bkw_ yeah it was fun. it's a dual brained setup, sip phone -> local asterisk pbx inside nat -> colo outside nat .. the colo outside nat terminates FWD and IAXTEL and all calls route into the sip phone |
| 00:24.47 | tclark | Mar 17 15:58:30 DEBUG[139282]: rtp.c:1003 ast_rtp_raw_write: Difference is 635946336, ms is 1531119464 |
| 00:24.47 | bkw_ | zaptel on bsd would be |
| 00:24.49 | bkw_ | but not asterisk |
| 00:25.23 | tclark | 1531119464ms on a 100ms ping end pt :) |
| 00:25.25 | tessier | Why bother? |
| 00:25.35 | todd- | someone call 17002279094 .. you'll either get me or an openbsd asterisk but you'll be talking through openbsd asterisk anyway |
| 00:25.45 | espen | bkw.. |
| 00:25.46 | espen | :) |
| 00:25.56 | bkw_ | todd asterisk on openbsd is not new |
| 00:26.06 | bkw_ | zaptel on openbsd would be orgasim worthy |
| 00:26.08 | todd- | working as an official port? |
| 00:26.21 | bkw_ | todd- you and tholo need to talk |
| 00:26.24 | killerbee | bkw_ good to CU now fix rtp.c |
| 00:26.30 | killerbee | lol |
| 00:26.33 | todd- | I know tholo, I'm his backup mx |
| 00:26.39 | bkw_ | killer rtp.c was still fucked as of today |
| 00:26.48 | bkw_ | one way works the other doesn't |
| 00:26.49 | tclark | as of now |
| 00:26.59 | bkw_ | tclark you still have problems with it? |
| 00:27.03 | rd | #6 0x292a6f58 in _init () |
| 00:27.03 | rd | <PROTECTED> |
| 00:27.03 | rd | #7 0x292a97ce in _init () |
| 00:27.03 | rd | <PROTECTED> |
| 00:27.03 | rd | #8 0x0805dae9 in pbx_exec (c=0x8148000, app=0x81085c0, data=0xbf9d8b68, |
| 00:27.03 | rd | <PROTECTED> |
| 00:27.04 | *** join/#asterisk daork (~nward@dhcp17.quicksilver.co.nz) |
| 00:27.05 | rd | #9 0x0805f76d in pbx_extension_helper (c=0x8148000, |
| 00:27.07 | rd | <PROTECTED> |
| 00:27.08 | tclark | Difference is 635946336, ms is 1531119464 |
| 00:27.09 | rd | <PROTECTED> |
| 00:27.11 | rd | <PROTECTED> |
| 00:27.12 | tclark | lol |
| 00:27.15 | bkw_ | rd so help me god |
| 00:27.20 | rd | that's a partial stack trace related to executing a dialer |
| 00:27.25 | bkw_ | 1 thru 5 is where the problem is |
| 00:27.33 | bkw_ | show us those |
| 00:28.01 | rd | #0 0x280f242f in pthread_testcancel () from /usr/lib/libpthread.so.1 |
| 00:28.01 | rd | #1 0x280ec703 in pthread_mutexattr_init () from /usr/lib/libpthread.so.1 |
| 00:28.01 | rd | #2 0x280eb2a9 in pthread_mutexattr_init () from /usr/lib/libpthread.so.1 |
| 00:28.01 | rd | #3 0x08135800 in ?? () |
| 00:28.01 | rd | #4 0x28228178 in adsi_transmit_message () |
| 00:28.02 | rd | <PROTECTED> |
| 00:28.04 | rd | #5 0x28228d9d in adsi_load_session () |
| 00:28.06 | rd | <PROTECTED> |
| 00:28.09 | bkw_ | wtf |
| 00:28.23 | espen | bkw.. msg.. |
| 00:28.24 | bkw_ | testcancel |
| 00:28.48 | bkw_ | looks like adsi is hozed |
| 00:29.02 | rd | bkw: I didn't show 'em since it's just a thread creation |
| 00:29.09 | bkw_ | so |
| 00:29.13 | bkw_ | thats where the problem starts |
| 00:29.30 | rd | bkw: so what I'm saying is that most likely my configuration is screwed up and I have no idea where to look |
| 00:29.31 | bkw_ | everythign in a bt is needed |
| 00:29.33 | *** join/#asterisk venix (~venix@CPE00400523b63e-CM000a7363ecf6.cpe.net.cable.rogers.com) |
| 00:29.36 | brif8 | ManxPower: brb. phone rang |
| 00:29.49 | bkw_ | rd do you have all the adsi stuffs in /etc/asterisk |
| 00:29.56 | bkw_ | or have you farked with it |
| 00:29.57 | *** join/#asterisk l-fy (~diana@home-25022.b.astral.ro) |
| 00:29.59 | bkw_ | ADSI should die |
| 00:30.00 | *** join/#asterisk alstor (alstor@shellshop.no) |
| 00:30.16 | rd | bkw: I don't believe so |
| 00:30.16 | bkw_ | DO NOT PRIVATE MESSAGE PEOPLE |
| 00:30.20 | bkw_ | its rude |
| 00:30.31 | bkw_ | i dont mind helping here in the channel |
| 00:30.35 | bkw_ | where everyone can learn from it |
| 00:30.42 | bkw_ | but one on one is 60.00/hr+ |
| 00:30.42 | denon | /msg bkw_ HELP! |
| 00:30.47 | bkw_ | denon haha |
| 00:30.48 | espen | we had a dialog.. |
| 00:30.55 | tessier | <PROTECTED> |
| 00:30.57 | denon | wtf is wrong with ADSI btw? |
| 00:31.04 | espen | bkw, well, kan you help me with my problem then? |
| 00:31.09 | espen | can. |
| 00:31.17 | tessier | denon: It's analog! |
| 00:31.17 | tessier | That is SO 1980's! |
| 00:31.22 | bkw_ | denon ADSI I have no use for it so it shouldn't be something that everything depends on |
| 00:31.40 | rd | bkw: I removed ADSI configs, since I didn't see them in the examples |
| 00:31.45 | bkw_ | the core of asterisk should be ripped and rebuilt with some sort of dependancy manager |
| 00:31.53 | alstor | bkw? Could you help me with my problem then? |
| 00:31.54 | bkw_ | rd put them back |
| 00:31.58 | bkw_ | alstor NO |
| 00:32.03 | alstor | im espen. |
| 00:32.04 | alstor | :P |
| 00:32.16 | rd | bkw: just adsi.conf or *.adsi as well? |
| 00:32.20 | bkw_ | rd yes |
| 00:32.32 | rd | bkw: yes to the first one or yes to both? |
| 00:32.36 | bkw_ | MustDie 2 hours min. |
| 00:32.43 | MustDie | shit |
| 00:32.43 | bkw_ | and 1 hour blocks there after |
| 00:32.45 | MustDie | i knew that |
| 00:33.02 | bkw_ | But I don't mind helping here in the channel at all |
| 00:33.08 | bkw_ | just most people refuse to read |
| 00:33.10 | bkw_ | or even think |
| 00:33.13 | alstor | bkw, why? we had a dialog on it.. |
| 00:33.21 | *** join/#asterisk heison (~heison@CPE000a01d49e6f-CM014300011132.cpe.net.cable.rogers.com) |
| 00:33.29 | bkw_ | and I'm not here to hold hands (granted it took me a week to get a hold of asterisk but I never asked stupid questions |
| 00:33.51 | jjhall | In extensions.conf, will SetCallerID(800-555-1212) work or do I need to do SetCallerID(${CIDNUM}) and set a globa for it? |
| 00:33.57 | rd | well, didn't help much |
| 00:33.58 | alstor | bkw, eh? ANSWER GODDAHM! |
| 00:34.06 | rd | let me see if stack trace changed |
| 00:34.15 | heison | bkw_: who is it this time? |
| 00:34.15 | *** join/#asterisk brc007 (~brc007@ip68-109-131-42.ph.ph.cox.net) |
| 00:34.20 | bkw_ | who what? |
| 00:34.38 | rd | #5 0x28228d9d in adsi_load_session () |
| 00:34.38 | rd | <PROTECTED> |
| 00:34.38 | rd | #6 0x292a6f58 in _init () |
| 00:34.38 | rd | <PROTECTED> |
| 00:34.38 | rd | #7 0x292a97ce in _init () |
| 00:34.39 | rd | <PROTECTED> |
| 00:34.39 | heison | bkw_: who is bothering you? for hand holding? |
| 00:34.41 | rd | #8 0x0805dae9 in pbx_exec (c=0x8149000, app=0x81085c0, data=0xbf9d8b68, |
| 00:34.42 | rd | <PROTECTED> |
| 00:34.45 | rd | #9 0x0805f76d in pbx_extension_helper (c=0x8149000, |
| 00:34.45 | *** join/#asterisk angler_ (~angler@24.214.255.57) |
| 00:34.46 | rd | <PROTECTED> |
| 00:34.48 | rd | <PROTECTED> |
| 00:34.50 | rd | <PROTECTED> |
| 00:34.52 | rd | no, it's pretty much the same |
| 00:35.01 | rd | I've added asterisk.adsi and adsi.conf |
| 00:35.20 | adelorenzo | I guess there is no suchthing as stupid question, and BTW btw_ sorry for pvt you earlier.... |
| 00:37.04 | *** join/#asterisk MamboKing (~mambo@d141-65-140.home.cgocable.net) |
| 00:37.10 | espen | OK, i have a problem with my cisco 7960. It will not work and it is acting crazy.. |
| 00:37.29 | killerbee | espen kick it.. then load a new image |
| 00:37.33 | heison | espen: what kinda problem? |
| 00:37.52 | espen | i un-plugged the power cord, when the phone was dialing |
| 00:38.04 | killerbee | crap.. how long had rtp been fucked |
| 00:38.31 | espen | suddently, and then it just rebooting.. after some hours it "worked" again, but would not connect to anything and hang alot.. |
| 00:38.49 | killerbee | espen what firmware you have? |
| 00:39.12 | espen | POS3-06-3-00 |
| 00:39.48 | killerbee | humm I only seem to have 6.2 |
| 00:39.53 | killerbee | THAT is a problem |
| 00:40.13 | rd | bkw: any other suggestions? |
| 00:40.22 | espen | problem for you or me, or both? :P |
| 00:40.27 | killerbee | somone send me 6.3 |
| 00:42.32 | *** join/#asterisk bobman (~bobman@68-169-206-81.agstme.adelphia.net) |
| 00:42.35 | espen | yes, send him that.. so he could help me.. |
| 00:42.36 | killerbee | espen seems both at this time |
| 00:43.15 | espen | hm? |
| 00:43.38 | killerbee | when did 6.3 come out? |
| 00:44.52 | espen | killerbee, but what could be the problem with my phone? |
| 00:45.09 | killerbee | dunno at this time.. reload 6.2 |
| 00:45.27 | espen | how? |
| 00:45.33 | espen | i dont have any files |
| 00:45.40 | killerbee | how did ya load 6.3? |
| 00:45.48 | rd | well, thanks whoever tried to help me |
| 00:45.50 | *** part/#asterisk rd (nobody@sivka.rdy.com) |
| 00:46.18 | espen | killerbee, from a tftp |
| 00:46.27 | MustDie | 6.3 is for MGCP |
| 00:46.33 | killerbee | did it ever work? |
| 00:46.34 | MustDie | not for sip |
| 00:46.41 | killerbee | oh... |
| 00:47.20 | killerbee | gots to run... |
| 00:47.24 | espen | killerbee, no |
| 00:47.46 | killerbee | espen I know nutting about mgcp... |
| 00:47.52 | killerbee | got the wrong guy ... |
| 00:47.53 | MustDie | 0.01 each |
| 00:47.57 | mitchel_ | can anyone here offer me a good outgoing voip -> land line rate? |
| 00:47.58 | espen | uh.. i want to use sip.. |
| 00:48.03 | espen | just want it to work again |
| 00:48.25 | MustDie | espen: do you have MGCP or SIP ? |
| 00:48.25 | slePP | mustdie: you paid $2.00 for the entire channel? cheapass. |
| 00:48.49 | espen | MustDie: Have, like; want to use? |
| 00:48.54 | LostFrog | Finally.. someone wants me.. |
| 00:48.57 | LostFrog | :) |
| 00:49.18 | tessier | I don't need a separate CSU/DSU if I am bringing in a data T-1 directly into an asterisk box do I? |
| 00:49.19 | espen | i have SIP-settings on it now.. |
| 00:49.21 | slePP | anyone know which codecs work off the 7910 (without me having to find the specs)? |
| 00:49.42 | espen | listen, the phone is fucking wierd.. newdial button is now there, etc, |
| 00:49.48 | espen | i not there |
| 00:49.48 | espen | even |
| 00:49.52 | espen | gah, is not there |
| 00:51.07 | brif8 | ManxPower: sorry got called away |
| 00:51.49 | adelorenzo | does anyone know how to force GSM on calls transfered from a Zap channel to a IAX /SIP? |
| 00:51.55 | brif8 | ManxPower: if I'm right you were saying the local loop and be Sprint while the LD and Local Service provider can be someone else ? |
| 00:52.06 | *** join/#asterisk Q-At-Home (~Queue@edtn007120.hs.telusplanet.net) |
| 00:52.12 | *** join/#asterisk Administrator (Administra@61.11.96.40) |
| 00:52.56 | Q-At-Home | anyone know if chan_iax2.c is buggered in latest cvs? |
| 00:52.56 | slePP | anyone know what model D-Link Primus gives you? |
| 00:53.03 | ManxPower | brif8: Yes, but you don't have to worry about that. The LD carrier will deal with your local telco for all that stuff. |
| 00:53.20 | ManxPower | You would call the LD carrier for repair on the local loop as well |
| 00:53.33 | ManxPower | They will coordinate with the local telco |
| 00:53.36 | *** join/#asterisk pooh__ (user30@cust.15.241.adsl.cistron.nl) |
| 00:54.24 | brif8 | ManxPower: ok cool, will they also handle 1-800 numbers and telephone number other than local, in other words I could get a NY and CA number even though I'm in FL ? |
| 00:54.31 | heison | ~seen jerjer |
| 00:54.32 | | jerjer <~NunYoBizN@dsl-69-29-10-120-grp.customer.centurytel.net> was last seen on IRC in channel #asterisk, 1d 20h 5m 47s ago, saying: 'put it in the [NuFone] type=peer'. |
| 00:55.22 | ManxPower | brif8: No. |
| 00:55.43 | ManxPower | brif8: The reasons are complicated, but basically you can assume no. |
| 00:56.06 | brif8 | ManxPower: ok how then can I get non-local numbers? |
| 00:56.10 | ManxPower | The LD carrier would also handle your inbound toll-free. |
| 00:56.36 | ManxPower | brif8: Before VoIP? You got a point-to-point T-1 line costing thousands of dollars per month. |
| 00:56.42 | lancey | anyone help me with my AS5350? |
| 00:56.53 | lancey | i can't figure out how to route calls to Asterisk |
| 00:56.58 | ManxPower | brif8: Basically getting numbers in remote locations is very expensive. |
| 00:57.31 | brif8 | ManxPower: ok with the 1800, I would still pay the 1c/min or whatever I contract to right |
| 00:58.01 | jjanzer | sweet i think I just fixed bug 1202 |
| 00:58.05 | brif8 | I was doing the maths while on the phone, I could prob. commit to 30 - 50 k mins / month |
| 00:58.16 | pooh_ | is there a way to test MOH ? (I'm by myself with no other users) |
| 00:58.34 | ManxPower | brif8: 1800 is a different service, and would be a different contract, but a decent LD carrier would allow you to send the toll free calls inbound via the T-1 they installed for your LD |
| 00:58.43 | tessier | pooh_: Just dial the moh extension |
| 00:59.28 | pooh_ | tessier: hmmm never thought of that (back to rtfm) thx |
| 00:59.32 | ManxPower | brif8: Keep in mind that many carriers won't let you add your inbound toll free and outbound LD to come up with a monthly comitment. It depends on the carrier. |
| 01:00.12 | *** part/#asterisk stonefly (~trillian@toby.stoneflytech.com) |
| 01:00.19 | brif8 | ManxPower: right, I was looking at 30 - 50 k just out bound (making LD, not rec. LD calls) |
| 01:00.31 | espen | 2004/03/18 01:59:30 UTC [2264/4048]: Processing TFTP request on interface 217.118.56.207... |
| 01:00.32 | espen | 2004/03/18 01:59:30 UTC [2264/4048]: Request from 217.118.49.203:50549 |
| 01:00.32 | espen | 2004/03/18 01:59:30 UTC [2264/2404]: Read request for cp7960/SIP000DED24D581.cnf; mode=octet |
| 01:00.32 | espen | 2004/03/18 01:59:30 UTC [2264/2404]: File not found. |
| 01:00.32 | pooh_ | tessier: uuhmm got an example of the MOH dialing syntax? |
| 01:00.36 | espen | which file should that be? |
| 01:01.02 | ManxPower | espen: That's the configuration file the phone expects. It's created by the Cisco call manager |
| 01:01.14 | brif8 | ManxPower: if I could get a rate like 1c/min based on that volume, then I could afford to cover the cost of all the inbound toll free calls |
| 01:01.27 | ManxPower | pooh_: exten => 666,1,Dial(Zap/1,20,m) |
| 01:01.48 | LostFrog | m=moh? |
| 01:01.49 | *** join/#asterisk mochouinard (~mochouina@modemcable105.167-203-24.mc.videotron.ca) |
| 01:01.52 | ManxPower | brif8: Talk to a couple of LD carriers. IDT has good residential rates, they might have good business rates |
| 01:02.01 | ManxPower | LostFrog: "show application dial" |
| 01:02.13 | brif8 | ManxPower: will have to search not heard the term IDT ? |
| 01:02.16 | LostFrog | lol.. thanks. |
| 01:02.26 | ManxPower | brif8: IDT is a discount Long Distance company. |
| 01:02.36 | ManxPower | brif8: I use them when not using NuFone. |
| 01:02.38 | *** join/#asterisk extremis (~extremis@207.235.121.17) |
| 01:02.44 | pooh_ | ManxPower: thx again |
| 01:03.09 | extremis | Has anyone experienced problems with calls dialing out a PRI not ringing, but give silence... and a redial will actually complete the call? |
| 01:03.18 | LostFrog | Cool. |
| 01:03.24 | ManxPower | brif8: Sprint, AT&T can get good rates too if you pressure them with quotes from cheaper LD companies. |
| 01:03.36 | ManxPower | extremis: Yes! |
| 01:03.39 | *** join/#asterisk Carp (Carp@ip-204-97-151-229.modem.logical.net) |
| 01:04.00 | brif8 | ManxPower: thank You !! |
| 01:04.04 | ManxPower | extremis: My carrier screwed up the configuration of one of our B-Channels |
| 01:04.22 | extremis | ManxPower: I put * in the middle of my PRI and my existing PBX and now people are complaining of this issue.... the PRI debug messages show the call being completed |
| 01:04.22 | ManxPower | extremis: Notice if happens on the same channel(s) every time. |
| 01:04.30 | *** part/#asterisk mitchel_ (~mitchel@216.132.54.140) |
| 01:04.33 | *** join/#asterisk cybershield (cybershiel@ppp-88-182.24-151.libero.it) |
| 01:04.50 | ManxPower | extremis: Could be your timeing source, could be your zap configs. How many channels do you have ACTIVE on your PRI? |
| 01:05.05 | extremis | active? |
| 01:05.06 | ManxPower | Make sure your switchtype is right. |
| 01:05.17 | extremis | switchtype was configured by kram |
| 01:05.22 | extremis | well, by me an kram |
| 01:05.27 | ManxPower | extremis: We have a PRI with channels 1-6 active, channels 7-23 unused and chan 23 for D channel. |
| 01:05.33 | ManxPower | Saves us a lot of money per month |
| 01:05.41 | extremis | well, 1-23 should be B |
| 01:05.44 | extremis | 24 D |
| 01:05.51 | extremis | or 1-22 and 23 |
| 01:05.53 | extremis | I forget |
| 01:06.06 | extremis | how many B channels does a PRI have? |
| 01:06.13 | Moc_ | 24 |
| 01:06.16 | LostFrog | Thanks ManxPower . |
| 01:06.43 | Moc_ | you can switch a B channel into a D channel at will |
| 01:06.49 | ManxPower | a {RI has @# b channels |
| 01:06.52 | ManxPower | eek |
| 01:06.59 | ManxPower | A PRI has 23 B channels and 1 Dchannel. |
| 01:07.05 | Moc_ | not necesarry |
| 01:07.07 | ManxPower | extremis: What are the span= lines in your /etc/zaptel.conf? |
| 01:07.28 | ManxPower | Moc: In the USA they generally do. Not all the B channels might be in use. |
| 01:07.55 | Moc_ | Like here we got 4 T1, 2 T1 with 24B channel and 2 T1 with 23 B Channel and 1 D channel |
| 01:08.11 | Moc_ | second D channel is our backup |
| 01:08.15 | extremis | ManxPower: maybe related... is it normal to see the idle B channels reset all at the same time on a regular basis (possibly random) |
| 01:08.42 | ManxPower | Moc: That's called NFTAS where one D channel is shared between more than one PRI. But Asterisk does not suppor NFTAS |
| 01:08.48 | ManxPower | extremis: Yes. |
| 01:08.56 | angler_ | extremis, hows it going? |
| 01:09.03 | extremis | howdy angler: pri problems |
| 01:09.07 | extremis | span=1,1,0,esf,b8zs |
| 01:09.07 | extremis | span=2,2,0,esf,b8zs |
| 01:09.11 | angler_ | ah whats wrong |
| 01:09.18 | Moc_ | I know, but you still dont need a D channel.. depending of the usage |
| 01:09.20 | *** join/#asterisk BoRiS2 (~boris@24.77.165.150) |
| 01:10.04 | extremis | people are reporting that when they make outgoing calls from the phones on the panasonic pbx out the pri to the * box out its pri to the telco (logix) that often the first call will sit in silence until they hang up, but if they redial it goes through fine |
| 01:10.17 | extremis | and I did pri debug span 1 and span 2, bur I can't see the cause |
| 01:10.52 | extremis | * sitting between the telco and the old PBX is temporary, so to prove that * is superior |
| 01:11.00 | *** join/#asterisk Poincare (~jeff@D577A8BF.kabel.telenet.be) |
| 01:11.04 | extremis | but they have all lsot faith since these odd issues started when I put the * box in |
| 01:11.28 | extremis | I was hoping to move completely to * , but this is holding us up |
| 01:11.40 | extremis | we can't seem to duplicate it |
| 01:11.44 | extremis | it just happens randomly |
| 01:12.17 | SimonR | Has anyone tried using the 2.6 kernel with Asterisk so it will work with hyperthreading and dual processors. |
| 01:12.21 | Moc_ | you defined your group to 24 line when you only have 24... I dont know |
| 01:12.39 | Moc_ | second 24 should be 23 |
| 01:13.19 | extremis | bchan=1-23,25-47 |
| 01:13.20 | extremis | dchan=24,48 |
| 01:13.26 | ManxPower | extremis: You also have bchan=1-23 dchan=24 |
| 01:13.37 | ManxPower | extremis: Ah, OK. Yes. |
| 01:13.42 | BoRiS2 | I have it work with 2.6 kernel |
| 01:14.16 | SimonR | is this so you can use a Xeon? |
| 01:14.42 | BoRiS2 | I have a hyperthreading cpu but haven't tried it on it |
| 01:15.06 | lancey | *ANYONE* here using Cisco AS5350? |
| 01:15.15 | lancey | i need some help, plz |
| 01:15.22 | extremis | switchtype=national |
| 01:15.22 | extremis | pridialplan=unknown |
| 01:15.43 | SimonR | Is there a point to using a high-end machine with Asterisk? |
| 01:15.50 | ManxPower | extremis: So the channels you have problems with are not always the same channels? |
| 01:15.51 | SimonR | Like with dual processors or a Xeon? |
| 01:16.09 | ManxPower | extremis: I assume you have pri_cpe for the telco side and pri_net for the PBX side? |
| 01:16.14 | pooh_ | ManxPower: exten => 666,1,Dial(IAX2/1,20,m) gives me a ringtone and hangs up on me after some time |
| 01:16.18 | extremis | ManxPower: yes, to the pri_net,cpe |
| 01:16.24 | ManxPower | also try setting span span=2,0,0,esf,b8zs |
| 01:16.27 | BoRiS2 | SimonR: just lots of ram :) |
| 01:16.28 | extremis | ManxPower: don't know to the same channel issue since its hard to trace |
| 01:16.38 | ManxPower | Asterisk may not PROVIDE timeing to the PBX if that span is a secondary timeing source. |
| 01:16.39 | jjhall | SimonR: More simultaneous calls while transcoding and/or MOH, etc... |
| 01:16.42 | extremis | I have to wait until they bitch and then grep through a log |
| 01:16.48 | extremis | and I wasn't logging well |
| 01:17.00 | extremis | since a historical log of pri debug span isn't possibly without attaching to the console |
| 01:17.12 | ManxPower | And I'll bet your PBX expects to get it's timeing from the "telco" |
| 01:17.22 | jjanzer | hey I know this sounds like a silly question, but is the current day's cvs decently stable, like no new bugs? wondering if I should update to the latest for my production box |
| 01:17.23 | extremis | the telco says that they do not offer timing |
| 01:17.37 | extremis | so I'm assuming the pbx that I put this in front of provides its own timing |
| 01:17.43 | ManxPower | extremis: I'll bet they lie. |
| 01:17.50 | extremis | ManxPower: I have no way to tell |
| 01:17.59 | ManxPower | extremis: Try it and see. 8-) |
| 01:18.13 | extremis | ManxPower: try what? |
| 01:18.27 | ManxPower | span=2,0,0,esf,b8zs |
| 01:18.27 | Q-At-Home | I cant compile chan_iax2 from todays cvs... but that could be my old ass cvs leftovers |
| 01:18.29 | Q-At-Home | :) |
| 01:18.43 | Q-At-Home | time to rm -rf and re checkout |
| 01:18.48 | extremis | what happens if it is wrong? |
| 01:18.52 | ManxPower | Since your span=1,1,0,esf,b8zs is telling Asterisk to get its timeing from the telco |
| 01:19.03 | Carp | anyone know where I can find the RPM for apache? Redhat 8 |
| 01:19.06 | ManxPower | extremis: You'll get similar issues to what you have now. |
| 01:19.19 | extremis | manxpower heh |
| 01:19.35 | extremis | would there be an error in the pri debug span? |
| 01:19.56 | ManxPower | extremis: I don't know. You would see errors using zttool |
| 01:20.15 | extremis | so, if it is misconfigured I should see the errors in zttool now |
| 01:20.15 | extremis | ? |
| 01:20.25 | ManxPower | Yes, if you have timeing issues. |
| 01:21.07 | ManxPower | On a channelized voice T-1 you would hear audio blips during a call when the frames slips. I don't know if that is the same for PRI |
| 01:21.13 | pooh_ | ManxPower: what's your URL again pls |
| 01:21.14 | jjanzer | Q-At-Home, heh |
| 01:21.18 | ManxPower | Useful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unofficial Links") and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/ |
| 01:21.32 | pooh_ | ManxPower, thx |
| 01:22.08 | jjhall | I wish I had a nickel each time I saw that list...LOL. Guess I can't say much, that is where I got my samples to start out with. :-) |
| 01:22.38 | *** part/#asterisk epreston (~epreston@Toronto-HSE-ppp3693092.sympatico.ca) |
| 01:23.15 | *** join/#asterisk Tangent (authdenied@connerdata-6.dsl.easynet.co.uk) |
| 01:23.24 | extremis | ManxPower: what errors should I see in zttool? |
| 01:23.26 | extremis | it shows green |
| 01:23.37 | Q-At-Home | has anyone seen jtodd around? I mailed him a few days ago about some new prompts... |
| 01:23.38 | *** join/#asterisk angler_ (~angler@24.214.255.57) |
| 01:23.43 | Q-At-Home | ~seen jtodd |
| 01:23.45 | | jtodd <~jtodd@ti.fox-den.com> was last seen on IRC in channel #asterisk, 15d 19h 11s ago, saying: 'Silly me, thinking that this was a much more complicated problem. :-)'. |
| 01:23.55 | Q-At-Home | egadz |
| 01:24.05 | ManxPower | extremis: pick the card then tab to SELECT and hit enter |
| 01:24.08 | *** join/#asterisk kilobit (~sam@adsl-203-123-17-230.qala.com.sg) |
| 01:24.24 | ManxPower | Look for Bipolar Viol |
| 01:24.32 | extremis | I have none |
| 01:24.41 | extremis | I do have 2 irq resets on both pris |
| 01:24.52 | extremis | but thats not it |
| 01:24.52 | ManxPower | extremis: That should not cause major issues |
| 01:24.54 | kilobit | is it possible to pass multiple arguments on agi? |
| 01:25.06 | ManxPower | Make sure cat /proc/interrupts shows the card on it's own IRQ |
| 01:25.07 | Q-At-Home | bbiab |
| 01:25.20 | extremis | ManxPower: it is |
| 01:25.22 | ManxPower | kilobit: Not until a VERY recent CVS. |
| 01:25.52 | kilobit | ManxPower awww.. ok..thanks.. |
| 01:26.00 | ManxPower | kilobit: I did AGI(/tmp/my.agi,arg1&arg2&arg3) and then parsed out the three args (from the single arg they came into the AGI as) |
| 01:26.20 | extremis | ManxPower: convinced its not clocking problems? |
| 01:26.50 | ManxPower | (my arg1, my arg2, my arg3) = split("&", $ARGV[0]); or something like that in perl |
| 01:26.53 | slePP | has anyone dealt with mgcp? |
| 01:27.14 | ManxPower | extremis: No, it still could be, but it's not terribly likley. |
| 01:27.22 | kilobit | ManxPower i also did that.. i split by ":" .. i just wondering if i did it in a wrong way.. |
| 01:27.44 | extremis | ManxPower: I confirmed that the old pbx used internal clocking |
| 01:27.52 | extremis | so I should use 0 for no clocking on both span right? |
| 01:27.56 | ManxPower | kilobit: I seem to recall a change in CVS that allowed passing more than one arg, but I don't remmeber when |
| 01:28.08 | ManxPower | extremis: Yes, I still recommend that. |
| 01:28.18 | kilobit | ManxPower ic.. thanks .. maybe ill try the cvs.. |
| 01:28.28 | ManxPower | OK, my first attempt at making humus is a failure. |
| 01:28.45 | brif8 | ManxPower: all the talk on T1 does the same work for OC-3 , by any chance ? |
| 01:29.01 | ManxPower | brif8: about your LD stuff? |
| 01:29.19 | extremis | ManxPower: so.... I'm not showing a channel go active in zttool while I make an inbound call |
| 01:29.27 | extremis | so I'm not too sure its accurate |
| 01:29.38 | brif8 | ManxPower: OC-3 for LD and Data, isn't 4 or 5 T1s = to OC-3 ? |
| 01:29.48 | ManxPower | Yes, it applies from the standpoint of LD, but since Asterisk does not support those types of lines you'd need something to split the OC-3 or DS-3 into multiple T-1s |
| 01:30.12 | BoRiS2 | OC1=55Mbps |
| 01:30.14 | ManxPower | OC-3 is 192Mbps, I think. divide that my 1.5 and you get the number of T-1s |
| 01:30.24 | ManxPower | DS-3 is 45Mbps |
| 01:30.41 | brif8 | ManxPower: ok great! and a PC can handle multiple TE410P cards right ? |
| 01:31.13 | ManxPower | brif8: Up to two cards according to Digium using dual xeon machine |
| 01:31.53 | ManxPower | well up to two T400P cards, you can prolly get more TE410P cards in a machine since they are supposed to use DMA |
| 01:32.07 | SimonR | Now, what kind of machine makes sense to use? Is it better to just use a low-end machine with 100 calls each, or to try to push it with more powerful systems? |
| 01:32.26 | ManxPower | SimonR: I would put 4 T-1 per machine. |
| 01:33.02 | ManxPower | and get a decent motherboard with one processor that can have a second CPU added to it. i.e. an SMP motherboard with 1 CPU |