| 00:00.56 | carrar | dandate2, 10megs |
| 00:01.22 | carrar | besafe |
| 00:01.44 | carrar | get 3 megs commit |
| 00:01.46 | WIMPy | Sounds like at least 3, but where can you get less than 100 in a datacenter? |
| 00:01.55 | carrar | burst to 10 |
| 00:02.20 | carrar | 30 g711 calls is about 2.6 megs |
| 00:02.46 | carrar | Are you handling RTP? |
| 00:02.48 | carrar | or just SIP? |
| 00:02.51 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 00:02.54 | carrar | if Just sip 1meg is fine |
| 00:03.48 | p3nguin | What makes it so I always have to manually join this channel even with auto-join enabled for it? |
| 00:03.53 | carrar | if the audio is passing through asterisk then 30 calls is really 60 |
| 00:04.05 | carrar | in which cast thats 5.2 megs |
| 00:04.09 | carrar | case |
| 00:04.23 | WIMPy | read about 38 |
| 00:04.58 | WIMPy | Or maybe 24 if reinvites can be used. |
| 00:05.10 | carrar | p3nguin, why do you keep joining this channel @!#$!@#! :) |
| 00:05.58 | WIMPy | p3nguin: Maybe you joining too early? |
| 00:13.12 | *** join/#asterisk Benwa (~Schnitzel@unaffiliated/benwa) |
| 00:14.34 | dandate2 | bandwidth calculator says 50 simultaneous calls requires 5mb |
| 00:14.52 | carrar | 50 what where |
| 00:14.54 | dandate2 | but im thinking this is cut in half if the caller is in queue right |
| 00:15.00 | dandate2 | im sorry 30 simultaenous calls |
| 00:15.00 | carrar | 50 in and 50 out? |
| 00:15.12 | dandate2 | 30 people calling the sales / customer support lines |
| 00:15.22 | dandate2 | and 6-8 people answering while the rest sit on hold listening to moh |
| 00:15.25 | carrar | Are the phone in the data center too? |
| 00:15.32 | dandate2 | no all remote |
| 00:15.45 | carrar | so thats really 2 channels per call then |
| 00:15.48 | carrar | not 1 |
| 00:15.58 | dandate2 | right |
| 00:16.05 | dandate2 | so would require 5mb no doubt right |
| 00:16.10 | carrar | and if they audio is passing through Asterisk |
| 00:16.19 | carrar | then 1 call is your codec * 2 |
| 00:16.44 | carrar | as far as bandwidthneeds |
| 00:16.52 | dandate2 | i mean 30 calls at once is like the maximum |
| 00:17.02 | dandate2 | ao i dont want to overshoot it |
| 00:17.03 | carrar | so thats 15 calls |
| 00:17.14 | dandate2 | i mean 30 people calling the sales line |
| 00:17.29 | carrar | or 30 concurrent calls is really 60 channels of bandwidth |
| 00:17.33 | dandate2 | right |
| 00:17.39 | WIMPy | sees 24 or 38 calls, dpending on setup. |
| 00:17.54 | dandate2 | but a concurrent call means someoen answered and 2 people are talking right |
| 00:18.13 | dandate2 | cuz as far as that goes, its only 6-8 concurrent calls |
| 00:18.24 | dandate2 | and up to 22-24 people sitting on hold |
| 00:18.33 | leifmadsen | don't worry about concurrent calls -- worry about concurrent channels |
| 00:18.35 | carrar | well if 10 people are on hold in a queue thats just 10 concurrent calls |
| 00:18.47 | leifmadsen | carrar: s/calls/channels |
| 00:18.50 | carrar | since they are not being passed to a remote phone |
| 00:18.57 | leifmadsen | a call is 2 channels |
| 00:19.01 | leifmadsen | (in most cases) |
| 00:19.08 | *** join/#asterisk Montys (~dmartinez@nat/digium/x-clvbdysgzmrnlxxf) |
| 00:19.14 | carrar | 2 channels if going back out of the server |
| 00:19.14 | leifmadsen | when talking about asterisk, a "call" is not really a term you need to use |
| 00:19.22 | carrar | with regard to bandwidth needed |
| 00:19.37 | leifmadsen | just talk about it in channels, it ends up being much less confusing |
| 00:19.55 | carrar | heh |
| 00:20.06 | carrar | just double everything and you are good |
| 00:20.21 | carrar | get 6 or 7 megs commit |
| 00:20.21 | leifmadsen | if you have a channel coming in, and a channel going out, that's 2 channels |
| 00:20.29 | carrar | burst to 10 or 100 |
| 00:20.44 | dandate2 | well i figured that a remote ip phone station can handle 6 calls with 512kb. so we'll double this and figure the pbx needs 1mb to handle 12 calls. 3mb to handle 36 |
| 00:20.48 | leifmadsen | don't worry about whether the channel coming in is associated with the channel going out (call) |
| 00:21.03 | carrar | dandate2, wrong math |
| 00:21.06 | dandate2 | sorry 1mb to handle 6 calls |
| 00:21.39 | dandate2 | 5mb to handle 30 |
| 00:21.46 | carrar | 5.2 |
| 00:21.56 | WIMPy | Talk in channels als leifmadsen said. Ad read my last comment as channels as well. |
| 00:22.05 | leifmadsen | you make it confusing when you talk about calls, but go ahead |
| 00:22.13 | leifmadsen | just worry about channels |
| 00:22.15 | carrar | 5.3 really |
| 00:22.35 | carrar | but you want to round up |
| 00:22.39 | carrar | to the nearest meg |
| 00:22.42 | dandate2 | im a telemarketer, i can only think in calls |
| 00:22.46 | carrar | if you want to cut it close |
| 00:22.49 | leifmadsen | 6 channels in plus 1 channel out, at 64kbps + 24kbps overhead = 88kbps * 7 = foo |
| 00:23.01 | leifmadsen | dandate2: then you stand to continue being confused and more complicated than necessary |
| 00:23.03 | leifmadsen | enjoy that |
| 00:23.19 | carrar | hahah |
| 00:23.29 | carrar | I use 88 as well |
| 00:23.29 | dandate2 | i know its a nightmare |
| 00:23.45 | dandate2 | but trial and error! we have come far |
| 00:23.50 | leifmadsen | the definition of insanity is continuing to do the same thing and expecting a different outcome |
| 00:24.08 | WIMPy | That whole conversation should have been two lines. |
| 00:24.12 | leifmadsen | no shit |
| 00:24.24 | leifmadsen | well, 3 |
| 00:24.30 | WIMPy | Nice one. |
| 00:25.00 | leifmadsen | "how much bandwidth do I need for X number of channels using G.711" |
| 00:25.03 | leifmadsen | "Y kbps" |
| 00:25.05 | leifmadsen | "thanks" |
| 00:25.20 | carrar | for a server in a datacenter |
| 00:25.25 | dandate2 | so you guys figure 88kb per call for the ip phone also? |
| 00:25.32 | dandate2 | the bandwidth calculator says 79 =/ |
| 00:25.40 | carrar | yes 88 |
| 00:25.44 | carrar | for g711 |
| 00:25.54 | leifmadsen | I'm using 64kbps (G.711) and 24kbps for IP etc overhead |
| 00:26.04 | dandate2 | ! |
| 00:26.08 | WIMPy | Well, actuially we don;t know if RTP could be diverted, so it might be 24 or 38 channels. |
| 00:26.10 | dandate2 | damn this bandwidth calculator |
| 00:26.43 | leifmadsen | you have to consider the bandwidth for SIP setup, and NOTIFY, etc... too in case you're using that, so that's why I pick 88kbps |
| 00:26.47 | dandate2 | i ordered up a 512kb T1 line for a remote location thinking it could support 6 ip phones but at 88kb i'll need 768kb t1 |
| 00:26.51 | leifmadsen | better to have too much DBR than too little |
| 00:26.55 | leifmadsen | (data bit rate) |
| 00:27.10 | leifmadsen | don't use G.711 then? |
| 00:27.13 | leifmadsen | G.729 |
| 00:27.30 | dandate2 | naw this is some fast paced hustling, we actually need god to help us with this sale |
| 00:27.33 | dandate2 | best not cut the quality heh |
| 00:28.21 | carrar | dandate2, and what if 1 phone has two calls? |
| 00:28.36 | carrar | are you planning for that? |
| 00:28.58 | *** join/#asterisk unspin (~unspin@S01060026f2f3042d.vc.shawcable.net) |
| 00:29.25 | *** join/#asterisk p3nguin_ (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 00:29.39 | *** join/#asterisk dandate3 (~gtejkgjke@124.106.46.100) |
| 00:29.46 | carrar | Is a full T1 really that more expensive? |
| 00:29.59 | dandate3 | <carrar> dandate2, and what if 1 phone has two calls? ::: naw they are hooked up to a queue |
| 00:30.20 | dandate3 | T1 here in the philippines is crazy expensive, $1300/month for 2MB |
| 00:30.35 | *** join/#asterisk ritztech (~ritztech@ip70-189-221-21.lv.lv.cox.net) |
| 00:30.54 | dandate3 | because the incumbant providers are charging the world 30 cents per minute to cal philippines, its illegal to have a PBX here |
| 00:32.08 | dandate3 | not that anyone gets in troubl for anything...they just drive up the cost of doing it |
| 00:33.21 | carrar | Probably shouldn't break the LAW |
| 00:36.03 | dandate3 | nah philippines has the lowest incarceration rate in the world |
| 00:36.16 | dandate3 | 24 people in prison per 100k population compared to 800 in america |
| 00:36.22 | dandate3 | there is no law heh |
| 00:36.51 | dandate3 | so the corporations have banded together to make the PBX impossible. the bandwidth is overpriced, and theres an hour long brown out each day to add some chargebacks to the equation if you were trying to compete |
| 00:37.51 | carrar | Sounds like a great place |
| 00:38.09 | dandate3 | and they dont mess around with chargebacks here...if you try to use a mastercard at a resteraunt they match your signature to that of your ID |
| 00:38.58 | dandate3 | no it really is a great place though if your american like me =P |
| 00:39.19 | dandate3 | you can support a family of 5 for $250/month with a 1st world standard of living |
| 00:39.42 | WIMPy | Just without INternet? |
| 00:39.50 | dandate3 | well you can get crap DSL for $20/mo heh |
| 00:40.03 | dandate3 | riddled with packetloss heh |
| 00:40.32 | WIMPy | Things that don't work, don't count. |
| 00:40.51 | dandate3 | ahh works good enough to not take a phone call =/, its really hit and miss sometimes its not bad |
| 00:41.12 | dandate3 | but mm the luxury |
| 00:41.27 | dandate3 | 4 bedroom 2 story modern house sells for like $20,000 |
| 00:41.42 | WIMPy | Maybe you should drop your current plans and open a wisp there? |
| 00:41.49 | dandate3 | wisp? |
| 00:42.03 | WIMPy | Wireless ISP |
| 00:42.04 | carrar | WIRELESS |
| 00:42.12 | dandate3 | bandwidth is too expensive |
| 00:42.26 | carrar | own it |
| 00:42.29 | dandate3 | sheesh i'd have to rent 2MB for $1300/mo |
| 00:42.42 | carrar | resell it |
| 00:42.51 | dandate3 | ah people here earn $2 / day |
| 00:42.52 | carrar | make chaChing!! |
| 00:43.03 | dandate3 | cant sell anything to philippinos heh |
| 00:43.06 | carrar | don't sell to them |
| 00:43.08 | dandate3 | bananas and coconuts |
| 00:43.36 | dandate3 | thats all there is! |
| 00:43.37 | carrar | so then thats just 650 customers |
| 00:43.40 | carrar | CAKE |
| 00:43.44 | WIMPy | Trade bits for bananas. |
| 00:43.55 | dandate3 | i see an american here like once per week |
| 00:43.59 | carrar | and you can provide a packet loss free network |
| 00:44.10 | dandate3 | that might be a lil tougher |
| 00:44.13 | dandate3 | with all the rain n stuff |
| 00:44.21 | carrar | make it happen |
| 00:44.25 | carrar | stop your excuses |
| 00:44.28 | *** join/#asterisk moy_ (~moy@74.12.99.29) |
| 00:44.34 | dandate3 | shh i might get deported if i step out of line heh |
| 00:45.06 | dandate3 | they need packetloss here, keeps the wages low |
| 00:45.08 | carrar | why do you care, you are already breaking the law |
| 00:45.33 | dandate3 | my call center workers are doing a job i used to get paid $200-$300 a day for...for $5 a day |
| 00:45.40 | dandate3 | because of the packet loss |
| 00:46.14 | carrar | yup and in time that will end |
| 00:46.29 | dandate3 | yeah it will, the land/housing price here is skyrocketing |
| 00:46.51 | dandate3 | middle class americans are fleeing here in droves |
| 00:47.12 | carrar | Hows that health care system? |
| 00:47.20 | dandate3 | its not bad really and very cheap |
| 00:47.30 | dandate3 | the schools here all preach in english |
| 00:47.31 | carrar | heh |
| 00:47.35 | dandate3 | everyone speaks english |
| 00:47.41 | carrar | I think I'll stay in America :) |
| 00:47.42 | dandate3 | greatest damn english speaking counrty on earth imho! |
| 00:48.03 | dandate3 | well u got to visit thats how u never go back |
| 00:48.29 | carrar | visiting Japan and expirencing their health care system was enough for me |
| 00:48.37 | dandate3 | u get that beach paradise, near slave labour, inexpensive prices |
| 00:48.56 | dandate3 | and hot women everywhere |
| 00:50.21 | dandate3 | besides noone really needs healthcare, just do what the ancestors did...rub some tobacco on it and have a drink |
| 00:53.32 | dandate3 | no level of pain is left uncured by drunkeness =P |
| 00:54.24 | carrar | yeah |
| 00:54.32 | carrar | everything changes when you have a kid |
| 00:55.13 | dandate3 | i say bring em to philippines just to get out of that damn american school system |
| 00:56.24 | dandate3 | 12 years education, 0 job skills acquired, 0 usable knowledge. better to just learn a trade at the age of 13 |
| 00:57.11 | carrar | Thats what parenting is for |
| 00:58.06 | dandate3 | damn straight |
| 00:59.23 | dandate3 | that school though is a waste of time, i remember in highschool they "attempted" to offer job skills classes. by the time i graduated all that was obsolete and i was behind in the times for not actually working it 18 hours a day 6 days a week |
| 01:00.54 | dandate3 | school wanted to pretend that cisco was going to hire an unlimmited amount of network technicians for the rest of our lives heh |
| 01:01.27 | carrar | College is pretty much required now anyways |
| 01:01.41 | carrar | so plan on 4 years of that |
| 01:02.38 | carrar | get running with kahn 1st!! |
| 01:03.06 | dandate3 | oh man college is fraud |
| 01:03.20 | dandate3 | charging you a hundred thousands of dollars to read books you can get done on your own heh |
| 01:03.23 | carrar | doesn't matter |
| 01:03.52 | dandate3 | and i used to work for a recruiting firm, talkign to people with masters degrees that couldnt even find a job |
| 01:04.00 | carrar | I rather have a doctor who paid to go to medical school then one who taught himself |
| 01:04.05 | carrar | heh |
| 01:04.07 | dandate3 | lol |
| 01:04.17 | dandate3 | well heed this self tought doctor |
| 01:04.26 | dandate3 | tobacco + liquor = best medicine |
| 01:04.47 | carrar | Are you a self perscribed gynecologist? |
| 01:04.49 | carrar | heh |
| 01:04.53 | dandate3 | lol |
| 01:05.06 | dandate3 | sure |
| 01:05.18 | dandate3 | woman says she got a yeast infection, well you better smoke it out baby |
| 01:05.44 | dandate3 | rub some of this liquor on it to disinfect |
| 01:08.03 | carrar | You'll love watching the Milton Friedman videos |
| 01:08.18 | *** join/#asterisk moy_ (~moy@74.12.102.35) |
| 01:08.31 | joako | I have a problem where 1 SIP peer goes offline and the moment I restart Asterisk -- be it 5 minutes or 5 hours after the peer becomes unreachable -- the peer becomes reachable again. How can I resolve this? |
| 01:08.58 | carrar | fix your nat issue? |
| 01:09.40 | joako | carrar, If it's a NAT issue why does it work, stop working for no reason I can explain, and when I restart asterisk it starts working again -- for a while. |
| 01:10.04 | carrar | it's a great answer for what little info you gave |
| 01:10.43 | dandate3 | could be the crap lossy connection of your SIP peer |
| 01:10.50 | carrar | yeah |
| 01:10.53 | dandate3 | by your english format im guessing india |
| 01:11.01 | carrar | is this phone in the phillipines? |
| 01:11.21 | carrar | pour liqure on it |
| 01:11.26 | dandate3 | cheers |
| 01:11.26 | carrar | blow smoke on it |
| 01:11.30 | joako | No its in the USA, as am I, and the SIP peer is the ISP I set ping monitoring and that has not shown any issues |
| 01:11.30 | dandate3 | lol |
| 01:11.53 | dandate3 | i dont believe ur in the USA dude |
| 01:12.52 | joako-nocloak | dandate3, whois me |
| 01:12.55 | ritztech | anyone know of a way to fix a DELAY on inbound FXO (signaling fxs-ks) this is LOCAL from a PBX not from the carrier |
| 01:12.59 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 01:13.36 | ritztech | i see Starting simple switch on 'DAHDI/6-1' INSTANT but then the system doesnt start the dialplan until 3 seconds later |
| 01:14.05 | ritztech | so its something on dahdi / maybe echo cancel but ive tried MANY different things |
| 01:14.28 | dandate3 | odesk.com |
| 01:15.06 | dandate3 | then u can smoke and drink while the hard working pakistanis handle it all for you |
| 01:16.11 | joako-nocloak | dandate3, Why can't I smoke while I handle it myself? |
| 01:16.27 | joako-nocloak | I just stopped asterisk, sleep 60, start asterisk again and [Oct 1 21:14:43] NOTICE[31378]: chan_sip.c:13493 handle_response_peerpoke: Peer 'trunk_1' is now Reachable. (16ms / 5000ms) |
| 01:16.38 | dandate3 | your damn question is so incoherent u must be drunk |
| 01:16.48 | dandate3 | the sip peer is your ISP? |
| 01:17.14 | joako-nocloak | dandate3, SIP peer goes down (according to asterisk) daily at random times, the solution is to restart asterisk |
| 01:17.26 | dandate3 | damn |
| 01:17.29 | dandate3 | pbxinaflash homie |
| 01:17.32 | dandate3 | never did me wrong |
| 01:17.38 | joako-nocloak | When this happens in sip debug I only see outbound messages, nothing inbound from that trunk |
| 01:18.01 | dandate3 | do u get the same problem with other trunks |
| 01:19.03 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 01:19.16 | joako-nocloak | dandate3, No. |
| 01:19.45 | dandate3 | must be that trunk then |
| 01:19.51 | dandate3 | tell the trunk provider to move out of india |
| 01:20.22 | dandate3 | they are prolly hacking sip accounts of unsecure pbx's and reselling it to you..pretending its a trunk |
| 01:21.14 | joako-nocloak | dandate3, I understand why you would think that, but unless I restart Asterisk the trunk does not come back online. If the trunk was down restarting Asterisk would have no effect |
| 01:22.02 | dandate3 | the indians must have a system where when they detect your asterisk comes online, that feeds you a new hacked sip account to use as a trunk |
| 01:22.05 | dandate3 | *shrug* |
| 01:22.15 | dandate3 | chargeback |
| 01:22.17 | joako-nocloak | In other words restarting Asterisk directly correlates with the trunk coming back online. |
| 01:22.42 | joako-nocloak | The provider is Windstream and the sip trunk is delivered via a T1 line they installed. I highly doubt your explaination is the correct one. |
| 01:22.53 | carrar | restart every 60 seconds |
| 01:23.42 | joako-nocloak | carrar, I think it would be more efficient to use tin cans and string |
| 01:24.17 | dandate3 | make it lossless, they will come |
| 01:25.53 | joako-nocloak | Thanks to everyone for those great suggestions. I have to go for now. |
| 01:26.17 | carrar | Thanks for providing detailed information! |
| 01:26.19 | carrar | NOT |
| 01:27.40 | *** join/#asterisk florz (~florz@2001:1a50:503c::1) |
| 01:28.55 | ritztech | anyone know how to quicken the Dahdi Inbound call |
| 01:29.11 | dandate3 | odesk |
| 01:29.19 | dandate3 | or i can refer you to a voip master...$5/hour |
| 01:29.58 | dandate3 | these pakistanis will read the whole damn 30,000 page manual for you |
| 01:30.17 | dandate3 | and all the bug fixes |
| 01:31.17 | dandate3 | no joke you really cant fix anything hardware related unless you built it yourself. best to leave it to the masters |
| 01:32.43 | dandate3 | they'll be like..theres a delay between circuit Y and circuit Z so 0101010010101010 later and its fixed |
| 01:34.35 | dandate3 | did u buy the fxo card on ebay? |
| 01:35.00 | dandate3 | cuz ebay is fraud, they'll ship you broken ass junk and win the chargeback just cuz u signed for delivery |
| 01:53.34 | ritztech | i think immediate=yes Kinda fixed it but now i get ss-noserver |
| 01:53.37 | ritztech | service |
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| 03:23.57 | ritztech | as you can see Theres my 4 second Delay |
| 03:23.58 | ritztech | [2010-10-01 20:18:06] VERBOSE[20853] chan_dahdi.c: -- Starting simple switch on 'DAHDI/7-1' |
| 03:23.58 | ritztech | [2010-10-01 20:18:10] WARNING[20853] chan_dahdi.c: Unable to enable echo cancellation on channel 7 (No such device) |
| 03:23.58 | ritztech | [2010-10-01 20:18:10] VERBOSE[20853] pbx.c: -- Executing [4001@from-zaptel:1] Set("DAHDI/7-1", "DID=4001") in new stack |
| 03:23.58 | ritztech | [2010-10-01 20:18:10] VERBOSE[20853] pbx.c: -- Executing [4001@from-zaptel:2] Goto("DAHDI/7-1", "s,1") in new stack |
| 03:24.40 | ritztech | i dont even have echo cancelers Is there a way to FULLY disable it ( i tried Doing all things with CID that didnt fix it_ |
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| 09:59.50 | *** join/#asterisk DelphiWorld (~Delphi@41.200.16.123) |
| 09:59.52 | DelphiWorld | hi all |
| 09:59.58 | DelphiWorld | anyone using IP0X apliance? |
| 10:04.50 | *** join/#asterisk dandate2 (~gtejkgjke@58.69.25.48) |
| 10:05.22 | dandate2 | Should I expect an IP phones bandwidth requirement (at the IP phones end) to require 79kb per call or 88kb? |
| 10:06.29 | *** join/#asterisk UQlev (~Yuriy@212.50.99.8) |
| 10:06.52 | Jasnejac | why that high? a single ISDN line is only 64k |
| 10:07.07 | dandate2 | IP overhead |
| 10:08.03 | dandate2 | <@leifmadsen> I'm using 64kbps (G.711) and 24kbps for IP etc overhead ::: is he talking about the PBX bandwidth requirement or the ip phone? |
| 10:11.24 | dandate2 | i'm watching a windows bandwidth monitor while making a call on eyebeam, it seems to fluctuate from 79kbps to 100 kbps both up and down at the same time, does this sound right? |
| 10:13.32 | *** join/#asterisk niekie (~niek@CAcert/Assurer/niekie) |
| 10:15.59 | dandate2 | gunna lose mirc to try to get a more acurate stat heh |
| 10:18.25 | DelphiWorld | anyone from china |
| 10:27.48 | *** join/#asterisk Faithful (~Faithful@180.194.0.71) |
| 10:31.56 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 10:40.04 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 10:40.21 | AliRezaTaleghani | hello |
| 10:41.03 | AliRezaTaleghani | i had a preblem with dialplans related to agents answer the queued call |
| 10:41.42 | AliRezaTaleghani | how call take my hand to work more? |
| 10:43.42 | DelphiWorld | need a chinese person |
| 11:01.13 | *** join/#asterisk pa (~pa@unaffiliated/pa) |
| 11:12.25 | *** part/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 11:20.22 | *** join/#asterisk dandate2 (~gtejkgjke@58.69.25.48) |
| 11:20.57 | dandate2 | will ip phones utilizing T1 that are remote from the pbx face ip overhead or would they use a consistent 64kb each |
| 11:42.43 | *** join/#asterisk BANSAL (~bansal@117.199.123.192) |
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| 11:50.11 | drmessano | dandate2, that's more like 85k in each direction |
| 11:53.18 | Jasnejac | dandate2: This helps http://www.voip-info.org/wiki/view/Bandwidth+consumption |
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| 12:42.22 | *** join/#asterisk AliRezaTaleghani (~AliRezaTa@unaffiliated/AliRezaTaleghani) |
| 12:42.31 | AliRezaTaleghani | hello |
| 12:42.42 | *** join/#asterisk ghenry (~ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 12:44.05 | AliRezaTaleghani | can someone help me about Callers whom are in a queue, and listening to MOH! |
| 12:44.28 | AliRezaTaleghani | i need to play something to the Callers, just as the time the Agent is answering them... |
| 12:44.33 | AliRezaTaleghani | how can i ? |
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| 13:38.01 | fors1 | just upgraded from 1.4 with zaptel to 1.6 with dahdi. works well, except that after a reboot I have to run "dahdi_cfg -vvv" before asterisk is able to load chan_dahdi.so. Why is that? And how can I do a workaround? This is a sangoma card, wanrouter is started before asterisk, and wanrouter status shows everything ok. But seems like the channels have to be initialized or something by dahdi_cfg before asterisk can use them |
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| 15:00.58 | *** join/#asterisk cusco_ (~trilili@213.63.137.210) |
| 15:00.59 | cusco_ | hi |
| 15:01.41 | cusco_ | using transfer from features.conf how does asterisk decide which is a valid or invalid extension to transfer to?? |
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| 15:34.47 | *** join/#asterisk rethus (~suther@p5087B152.dip.t-dialin.net) |
| 15:35.21 | rethus | if i set callerid with an agi-script. how can i see this callerid normaly? only with meetme list <conference> , right? |
| 15:43.47 | cusco_ | what do you mean see caller id normally? |
| 15:47.58 | Kobaz | rethus: you're mixing all kinds of things |
| 15:49.38 | DelphiWorld | anyone using ip0X? switchfin |
| 16:04.24 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
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| 16:40.00 | Kobaz | so hmm... i'm having dtmf problems... all endpoints/peers/etc are set up for rfc2833, i send digits over a somewhat crappy connection... half the digits are not recieved, and some are recieved wrong |
| 16:40.14 | Kobaz | for instance i'll send '8000', and the remote side will get '880' |
| 16:48.32 | *** join/#asterisk [cannibalera] (~cannibale@201-2-232-115.fnsce703.dsl.brasiltelecom.net.br) |
| 16:50.46 | *** join/#asterisk rethus1 (~suther@p5087E0EA.dip.t-dialin.net) |
| 16:50.51 | rethus1 | sorry, my messenger ton't make a "Beep" :) so i didn't see, that you answer |
| 16:52.29 | rethus1 | i use a agi-script to manage my incomming calls... and i see a incomming call with konfernece list (instead of meetme) |
| 16:53.07 | rethus1 | but i have the problem, that i don't can set a callerid in this channel if i list it |
| 16:53.13 | rethus1 | you know what i mean |
| 16:53.48 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 16:54.55 | *** join/#asterisk Tim_Toady (~moi@188.4.25.172.dsl.dyn.forthnet.gr) |
| 16:55.05 | WIMPy | Do I get it right, that UDPTL is something that's only used for T.38? |
| 16:58.39 | *** part/#asterisk rethus1 (~suther@p5087E0EA.dip.t-dialin.net) |
| 16:58.49 | *** join/#asterisk rethus1 (~suther@p5087E0EA.dip.t-dialin.net) |
| 17:01.30 | rethus1 | someone using konference instead of meetme? |
| 17:02.02 | [TK]D-Fender | retuSHOW us your callerid problem. Your descriptions on this (which has been a long while now) suck so far. |
| 17:03.57 | rethus1 | ok, i describe it as good as i can: |
| 17:03.58 | rethus1 | if a user callin via Twinkle (Sip-client over IP), a agi-script (php) is called, which try to set a callerid. I try to set the callerid, cause i need to get all the user into tha actual channel. So if i do "konference list kerpen ", i got this: (image follow) |
| 17:04.10 | cusco_ | hi.. I don't understand why our dahdi box, is converting to slin.... |
| 17:04.12 | cusco_ | :/ |
| 17:04.22 | cusco_ | what would be most common reason? |
| 17:04.55 | [TK]D-Fender | cusco_: Maybe you could describe (SHOW) us the REST OF THE CALL |
| 17:05.11 | cusco_ | lol |
| 17:05.20 | Kobaz | ooo |
| 17:05.28 | Kobaz | [TK]D-Fender: i know you're all up on dtmf problems |
| 17:05.44 | Kobaz | lemme get some logs |
| 17:06.28 | [TK]D-Fender | Kobaz: You shouldn't get double DTMF on RFC2833, but could lose digits on packet loss. Actually... doubled could maybe happen if your packets are arrviing out of order. |
| 17:06.45 | rethus1 | http://yfrog.com/cbasteriskrj |
| 17:06.52 | [TK]D-Fender | Kobaz: And I'm not really an expert... I know a little bit ehre and there on this really... |
| 17:07.40 | Kobaz | mm |
| 17:07.51 | Kobaz | [TK]D-Fender: what's a good way to guarantee dtmf delivery then? |
| 17:07.58 | rethus1 | so user is only a id. Before, i remember - in meetme - i could set it to Username(phone-number) |
| 17:07.59 | Kobaz | would sip info be a better choice? |
| 17:07.59 | [TK]D-Fender | rethus1: I don't even see where that command is SUPOPSED to show you a callerID. I do not see you setting one. I do not see what the call had when it came IN in the first place. |
| 17:08.17 | WIMPy | Kobaz: A reliable network connection. |
| 17:08.21 | Kobaz | haha |
| 17:08.26 | Kobaz | hahaha |
| 17:08.28 | [TK]D-Fender | Kobaz: What is on the other end? |
| 17:08.52 | Kobaz | [TK]D-Fender: there's a bunch of links in the chain... i'm trying to find the source of the problem |
| 17:09.07 | Kobaz | but it's: sip endpoint, asterisk, sip->t1, t1->sip, asterisk |
| 17:09.11 | carrar | nice who is kojiharunyan |
| 17:09.15 | carrar | she's hot |
| 17:09.21 | [TK]D-Fender | Kobaz: What UA's <--- |
| 17:09.38 | *** join/#asterisk guax (~guax@unaffiliated/guaxinim) |
| 17:09.41 | Kobaz | agents: polycom, asterisk, adtran 904, adtran 904, asterisk |
| 17:09.50 | guax | anyone know if inter-asterisk pickup is possible? |
| 17:10.07 | Kobaz | [TK]D-Fender: i can go directly to the far asterisk with sip using a softphone, and digits work fine |
| 17:10.20 | Kobaz | [TK]D-Fender: so i have a feeling it's less of a networking problem and more of just, something is botched somewhere |
| 17:10.57 | [TK]D-Fender | Kobaz: You know you are telling me LESS as you go now... |
| 17:11.12 | carrar | 30 seconds, pls fix |
| 17:11.12 | Kobaz | hm... that's the setup |
| 17:11.34 | Kobaz | i'll paste up the rtp logs |
| 17:11.38 | cusco_ | [TK]D-Fender: I noticed it is only when dialing the call to another * box with g729 codec. Nativeformat is g729 but Write and Read are slin... |
| 17:11.47 | Kobaz | carrar: me love you long time |
| 17:11.54 | cusco_ | I also read on the web that in order for asterisk to detect silence on g729 it must transcode to slin... |
| 17:12.07 | cusco_ | how do I disable silence detection? |
| 17:12.12 | Kobaz | [TK]D-Fender: where's your less coming from? |
| 17:12.23 | carrar | maxsilence = 0 |
| 17:12.36 | [TK]D-Fender | cusco_: SLN is the version transcoding uses as the "lowest common denominator" |
| 17:12.43 | Kobaz | [TK]D-Fender: i really don't know much more, i'm trying to figure out which link is the problem |
| 17:12.50 | [TK]D-Fender | Kobaz: You're getting more vague instead of more specific |
| 17:12.58 | cusco_ | carrar: in sip.conf ? |
| 17:13.10 | carrar | voicemail.conf |
| 17:13.11 | Kobaz | i don't have specifics yet, i'm weeding out possible sources right now |
| 17:13.14 | cusco_ | [TK]D-Fender: that means that its not has heavy transcoding? |
| 17:13.21 | rethus1 | <PROTECTED> |
| 17:13.21 | rethus1 | User #: 01 <unknown> <no name> Channel: Console/dsp (unmonitored) 00:00:13 |
| 17:13.37 | cusco_ | carrar: but we do not use voicemail... do i still set it? |
| 17:13.40 | [TK]D-Fender | cusco_: It likely means it is DOING transcoding. Who said anything about "heavy"? |
| 17:14.05 | rethus1 | so if i use setcallerid... <unknown><noname> will be replaced. But it seems not to be possible to do this in konference |
| 17:14.21 | carrar | cusco, perhaps someplace else you are missing a 729 setting |
| 17:14.21 | [TK]D-Fender | rethus1: Why are we comparing MeetMe with Konference? They are 2 completely different apps, no? |
| 17:14.23 | cusco_ | yea, wel.. |
| 17:14.47 | Kobaz | i hate dtmf |
| 17:14.48 | rethus1 | i compare to show you, what i'm talking about! |
| 17:14.52 | [TK]D-Fender | rethus1: SHOW ME the complete call. |
| 17:15.10 | [TK]D-Fender | rethus1: those apps are DIFFERENT. They don't work the same. Stop expecting them to. |
| 17:15.36 | rethus1 | i know this. my question is: Is there a way to set a CallerID in Konference... thats all |
| 17:15.41 | bougyman | Kobaz: take it one leg at a time. |
| 17:15.56 | bougyman | put a dtmf echo app on the first asterisk box, test the polycom to it. |
| 17:16.07 | Kobaz | yeah |
| 17:16.09 | Kobaz | the polycom works fine |
| 17:16.10 | bougyman | i guess you can't put one on the adtrans but then try the same app on the far end asterisk box. |
| 17:16.13 | Kobaz | the local asterisk works fine |
| 17:16.13 | [TK]D-Fender | I don't give a shit that a Ford F-350 can tow a yaght when you actual own a Toyota Echo which can barely carrie GROCERIES |
| 17:16.15 | bougyman | well what doesn't work? |
| 17:16.27 | Kobaz | the remote asterisk works fine |
| 17:16.30 | carrar | F-450!! |
| 17:16.32 | Kobaz | it's something to do with the adtran to adtran |
| 17:16.33 | [TK]D-Fender | [13:15]<rethus1>i know this. my question is: Is there a way to set a CallerID in Konference... thats all <--- Konference doesn't SET CALLERID. |
| 17:16.48 | Kobaz | bougyman: passing dtmf over the pstn is getting botched |
| 17:16.50 | [TK]D-Fender | rethus1: "core show function CALLERID" <--- THIS sets CallerID |
| 17:16.58 | Kobaz | bougyman: is my theory thus far |
| 17:16.59 | *** join/#asterisk path (~path@gateway/shell/bshellz.net/x-powkutvabrvorvxa) |
| 17:17.03 | path | hello guys |
| 17:17.10 | bougyman | Kobaz: unlikely. |
| 17:17.16 | [TK]D-Fender | rethus1: You are even failing to show me where Konfernce gives a hshit about callerID or even DISPLAYS it so far. |
| 17:17.16 | path | is it mandatory to rebuild * if I want to upgrade my libpri ? |
| 17:17.17 | Kobaz | my thoughts too |
| 17:17.27 | bougyman | can you call from the pstn to one or the other asterisk boxes? |
| 17:17.31 | [TK]D-Fender | path: No. |
| 17:17.33 | rethus1 | i need not to make konference able to set it, i need only too output in in the Konference List <hchannel> |
| 17:17.37 | Kobaz | bougyman: yeah, doing some more testing |
| 17:17.41 | bougyman | if one works and the other doesn't you may find a setting of in an adtran. |
| 17:17.45 | path | [TK]D-Fender: thanks |
| 17:17.48 | bougyman | have you checked their settings? |
| 17:18.04 | [TK]D-Fender | rethus1: Well maybe that command DOESN'T do it, and CAN'T. It wasn't MADE to do this |
| 17:18.19 | Kobaz | all the settings looks fine.. although there really isn't much to control dtmf on for the adtrans |
| 17:18.20 | [TK]D-Fender | rethus1: You are asking about things that clearly don't seem to exist. |
| 17:18.41 | bougyman | Kobaz: stuff like rtp packet settings could affect it. |
| 17:18.54 | bougyman | you said T1 is the media, right? |
| 17:18.57 | Kobaz | bougyman: it's really weird... like... i can call tellme.. and it gets my digits just fine |
| 17:19.00 | Kobaz | yeah |
| 17:19.07 | cusco_ | I still can't figure why it is transcoding. [TK]D-Fender would a sip trace help? |
| 17:19.17 | carrar | did you set that? |
| 17:19.38 | [TK]D-Fender | cusco_: Your description of the COMPLETE call is VERY weak. Fix this. |
| 17:20.04 | cusco_ | lol |
| 17:20.04 | cusco_ | ok |
| 17:20.23 | carrar | also make sure you are not using any monitorign in your dial statements |
| 17:21.00 | cusco_ | err.. I am mixmonitoring the calls at the gateway |
| 17:21.20 | cusco_ | before the dial |
| 17:21.35 | cusco_ | ok let me paste dialplan, peer info and sip trace |
| 17:21.54 | [TK]D-Fender | GAH |
| 17:22.15 | cusco_ | that is not enough? |
| 17:22.46 | [TK]D-Fender | cusco_: Jesus fucking Christ..... Just tell me what fucking DEVICE is on each side and the damn codecs involved. If you are recording, Spy-ing, or MeetMe'd in any way |
| 17:23.52 | cusco_ | incoming from pstn --> astbox1 with dahdi and mixmonitoring --> SIP dial with peer using only g729 to ast-box2 --> dial to another peer |
| 17:24.48 | cusco_ | mixmonitor records in gsm |
| 17:24.55 | cusco_ | is that the reason...? |
| 17:25.04 | Kobaz | bougyman: ah, it looks like the local adtran (or something behind it) is affecting reliable delivery of dtmf |
| 17:25.16 | bougyman | there ya go |
| 17:25.32 | Kobaz | bougyman: i set up a Read on a straight dahdi system with a pri |
| 17:25.42 | Kobaz | sometimes i get 8000, sometimes i get 80, sometimes i get 800 |
| 17:25.53 | Kobaz | i haven't seen duplicates though |
| 17:25.53 | p3nguin_ | cusco_: MixMonitor records in the format you tell it to record in. |
| 17:26.14 | cusco_ | p3nguin_: which is gsm |
| 17:26.30 | carrar | You should transcode that call a few more times |
| 17:26.33 | [TK]D-Fender | cusco_: You asked why it hits SLN. Well your first box takes in DAHDI, and spits out G729 to your other box. It is also RECORDING. There goes 2 Encode, and 2 decode licenses, and it will go through SLN. Got another question? |
| 17:27.27 | cusco_ | why will it go trough SLN? |
| 17:27.52 | cusco_ | to save up licenses? |
| 17:28.09 | [TK]D-Fender | cusco_: No, because that is what TRANSCODING DOES |
| 17:28.29 | cusco_ | I thought it would transcode to gsm to record and to g729 to go out |
| 17:28.36 | path | what could possible mean this? http://pastebin.com/v0dDGXBt |
| 17:28.42 | cusco_ | ok so that should be normal... |
| 17:28.44 | path | is it not signaling properly? |
| 17:28.46 | [TK]D-Fender | cusco_: It does. 2 transcodes from the DAHDI side |
| 17:29.23 | [TK]D-Fender | path: Clocking or interrupt issue |
| 17:29.49 | Kobaz | bougyman: okay... adtran is doing fine... i can connect directly to it and make a sip call, and it sends proper dtmf every single time |
| 17:30.52 | path | [TK]D-Fender: from my * side or ISP ? |
| 17:31.42 | *** join/#asterisk Tim_Toady (~moi@188.4.144.62.dsl.dyn.forthnet.gr) |
| 17:34.17 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 17:35.18 | *** join/#asterisk RypPn (~TuMbL@rosscom.co.uk) |
| 17:36.51 | carrar | probably your asterisk box is miss configured |
| 17:37.10 | carrar | who knows |
| 17:37.16 | p3nguin_ | It could be mister configured. |
| 17:37.21 | [TK]D-Fender | path: ISP? that is a PRI. that isn't DATA |
| 17:37.22 | carrar | heh |
| 17:37.31 | [TK]D-Fender | path: try again... |
| 17:37.49 | [TK]D-Fender | path: And this is your CARD. |
| 17:39.17 | carrar | it's in a server he doesn't have access to since he didn't post any configs |
| 17:39.18 | path | eheh I meant my pri provider :) |
| 17:39.34 | carrar | Sure it's a PRI? |
| 17:42.14 | rethus1 | [TK]D-Fender: rethus1: You are even failing to show me where Konfernce gives a hshit about callerID or even DISPLAYS it so far.... |
| 17:42.15 | rethus1 | konference didn't do, but meetme does. |
| 17:42.41 | Maliuta | can someone confirm for me that in AEL I should be closing things out with just } and not }; ?? |
| 17:43.07 | path | http://pastebin.com/YfVPNkpp thats working |
| 17:43.15 | path | but this one not http://pastebin.com/81GG10WK |
| 17:43.29 | path | one is zaptel, the last one dahdi |
| 17:43.32 | carrar | open a Konfernce supprt ticket |
| 17:45.08 | path | the thing I got to isdn E1 trunks, but the second one goes up/down its flapping |
| 17:45.21 | path | s/to/two |
| 17:45.36 | path | both E1 trunks are the same supposedly |
| 17:46.14 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 17:46.16 | [sr] | hi :P |
| 17:46.46 | *** part/#asterisk guax (~guax@unaffiliated/guaxinim) |
| 17:46.51 | WIMPy | lo |
| 17:47.35 | [sr] | hi wierdo |
| 17:47.36 | [sr] | ops |
| 17:47.38 | [sr] | WIMPy: |
| 17:47.51 | WIMPy | LOL |
| 17:48.58 | WIMPy | [sr]: Do you still have contact to Andreas? I think he could really have a bug tracker. |
| 17:49.05 | [TK]D-Fender | [13:42]<rethus1>konference didn't do, but meetme does. <--- AND? Konference doesn't do it. THE END. |
| 17:49.21 | [sr] | WIMPy: he didn't even answered me :( i sent the same email twice.. |
| 17:49.58 | rethus1 | i don't can use meetme, cause i need a high performance konference-app |
| 17:50.10 | [sr] | WIMPy: i even sayd i would host it and maintain it, he just had to create an alias and point it to me from the misdn.org domain, but no answer.. |
| 17:50.21 | [TK]D-Fender | rethus1: So use it. But that command does not show you caller id. TFB <---- |
| 17:50.23 | rethus1 | now i search for a solution, to identify each incomming call exactly via this conference |
| 17:50.53 | WIMPy | [sr]: I didn't get a relpy to my last patch, either :-( |
| 17:51.10 | [TK]D-Fender | rethus1: You aren't even looking at what it DOES tell you. |
| 17:51.15 | [sr] | WIMPy: i'll try again but now to his personal email.. |
| 17:51.18 | [sr] | i'll let you know. |
| 17:52.12 | Maliuta | gah! turns out my problem was that "local" seems to be a reserved word in AEL |
| 17:52.34 | rethus1 | what did it tell me? |
| 17:52.44 | WIMPy | [sr]: I know someone who has better contact, I think. Maybe I can activate that route. |
| 17:52.59 | [sr] | WIMPy: please to that bridge and let me know. |
| 17:54.10 | [TK]D-Fender | rethus1: You tell ME. What does that command show you that could be used for this? |
| 17:54.51 | rethus1 | Yes, but i also tell you, that i can't use meetme |
| 17:55.05 | rethus1 | i need konference, cause its highperformance |
| 17:55.28 | [TK]D-Fender | rethus1: http://yfrog.com/cbasteriskrj <-- wake the fuck up and tell me what it DOES tell you here so you can get the fucking CALLERID |
| 17:56.18 | rethus1 | where is the callerid, please show me |
| 17:56.23 | carrar | hahah |
| 17:56.41 | carrar | You need to spend more time with Asterisk |
| 17:56.44 | carrar | and learn it rethus1 |
| 17:56.46 | [TK]D-Fender | rethus1: It isn't there. What IS there? |
| 17:57.02 | rethus1 | The column user have only a id, thats right... but if u had listen me, you would know, that i like to set NAME(NUMBER)! |
| 17:58.39 | [TK]D-Fender | rethus1: That command will NTO tell you the callerid. You can't do this in ONE FUCKING STEP. Just how dumb are you? Use that command. It will give you the CHANNEL NAMES. Go and FUCKING LOOK THEM UP WITH AMI and you get the fucking callerid! |
| 17:59.06 | [TK]D-Fender | God the idiocy level of this channel SKYROCKETS on weekends. |
| 17:59.27 | rethus1 | ha ha ha, [TK]D-Fender you realy funny. |
| 17:59.42 | [TK]D-Fender | rethus1: "konference list" gives you the CHANNELS. LOOK THEM UP. |
| 18:00.28 | carrar | 2 steps is IMPOSSIBLE!! |
| 18:01.39 | rethus1 | i talking about konference list <channel> not abotu konference list... keep on the watch... your blood pressure |
| 18:02.31 | [TK]D-Fender | rethus1: http://yfrog.com/cbasteriskrj <- this shows you the fucking CHANNEL NAME HERE. You can look it up and get the fucking callerid from it in ANOTHER STEP |
| 18:02.38 | carrar | You playing anywhere this weekend TK? |
| 18:02.41 | [TK]D-Fender | rethus1: SIP/dev-000....... |
| 18:03.17 | Kobaz | bougyman: interesting... i updated the firmware on the polycom.. i was running a semi old (1 year old) firmware. Now dtmf is working much better |
| 18:03.19 | rethus1 | this sended output is the result of konference list <myChannel> - relay! i don't lie |
| 18:03.24 | [TK]D-Fender | carrar: Nope.... I opted out of that new band I joined after my other one downsized |
| 18:03.46 | carrar | ah |
| 18:03.49 | [TK]D-Fender | rethus1: Look at your fucking pastebin. the ASTERISK channel is ther. USE it to get the callerid. |
| 18:03.57 | carrar | Does the new band offer better video? :) |
| 18:04.03 | rethus1 | and if it so, whay there is another channel in my channel? |
| 18:04.07 | rethus1 | i don't understand this |
| 18:04.14 | [TK]D-Fender | rethus1: SIP/dev-XXXX. OPEN YOUR FUCKING EYES. You can use this to GET the callerID |
| 18:04.35 | rethus1 | ok, I'll try it. |
| 18:04.42 | carrar | hire a Asterisk admin |
| 18:04.44 | [TK]D-Fender | rethus1: that tells you what CHANNEl is that particular user in the konference |
| 18:05.03 | carrar | read some books! |
| 18:05.15 | [TK]D-Fender | carrar: No amount of reading will increase his IQ. |
| 18:05.20 | carrar | heh |
| 18:05.28 | [TK]D-Fender | I suggest..... |
| 18:05.31 | rethus1 | thanks that you shout at me, and curse me. maybe it help me a bit ;) |
| 18:05.32 | [TK]D-Fender | ~osmosis |
| 18:05.32 | infobot | [~osmosis] Osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 18:05.33 | carrar | I TALKING THERE IS NO CALLER ID!! |
| 18:05.56 | rethus1 | i realy didn't realized, that there would be another channel insiede... but i 'll try this |
| 18:06.02 | [TK]D-Fender | And for those who don't follow, I'm hoping for the LATTER |
| 18:06.36 | [TK]D-Fender | rethus1: "another channel"? Holy shit. CHANNELS call Koference to JOIN the Konference. Channel = CALL. |
| 18:06.40 | [TK]D-Fender | NOT TO FUCKING BRIGHT |
| 18:06.46 | carrar | is that like channel vlans? ) |
| 18:07.33 | [TK]D-Fender | carrar: I could send him to his own :) |
| 18:07.47 | [TK]D-Fender | carrar: but that would violate my standards... |
| 18:07.53 | carrar | using g.711 encapsulation to to vlan that channel since of a conference channel! |
| 18:08.00 | carrar | since=inside |
| 18:08.02 | carrar | heh |
| 18:08.25 | carrar | my english is almost as bad as his |
| 18:13.59 | ChannelZ | me so solly |
| 18:14.05 | rethus1 | [TK]D-Fender: k, now i compare the different between channel and conferences... thanks for helping |
| 18:16.52 | path | if I want to upgrade dahdi (linux/tools) do I need to rebuild asterisk? |
| 18:17.25 | ChannelZ | generally depending on how far you're going |
| 18:17.58 | path | er what means how far |
| 18:18.03 | path | Im not going anywhere |
| 18:18.32 | ChannelZ | The version jump. |
| 18:21.01 | WIMPy | Quite interesting what prefixes the chinese Bots try out. |
| 18:22.25 | WIMPy | 009, 00011, 99559, 777, 9, 1, 0000, 00000. |
| 18:23.38 | ChannelZ | me chinese me play joke |
| 18:25.41 | *** part/#asterisk rethus1 (~suther@p5087E0EA.dip.t-dialin.net) |
| 18:26.29 | WIMPy | Is it a secret code? |
| 18:27.00 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 18:51.10 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 18:52.46 | DelphiWorld | anyone can give me example pathern for routing incomming call? |
| 18:53.20 | *** join/#asterisk diegoviola (~diego@190.23.48.144) |
| 18:53.24 | *** part/#asterisk diegoviola (~diego@190.23.48.144) |
| 19:02.43 | [TK]D-Fender | DelphiWorld: ... |
| 19:02.45 | [TK]D-Fender | ~book |
| 19:02.45 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 19:02.47 | [TK]D-Fender | ^^^^ |
| 19:02.53 | [TK]D-Fender | DelphiWorld: Chapter 5 is your friend |
| 19:03.06 | DelphiWorld | [TK]D-Fender: you are 30min old |
| 19:03.55 | [TK]D-Fender | DelphiWorld: No, I was in here last week... that makes me at LEAST 1 week old |
| 19:04.09 | [TK]D-Fender | DelphiWorld: And if you check further back you may find that I'm older still... |
| 19:04.14 | DelphiWorld | [TK]D-Fender: lol.. i mean i have found it allready and my call is working |
| 19:05.35 | DelphiWorld | [TK]D-Fender: one question please may by anoying... how do i route all 00 calls to a special trunk, i mean only the pathern. |
| 19:07.17 | [TK]D-Fender | DelphiWorld: go make a pattern to match that number... and do whatever you want with it. And never ever use the word "route" in here like that. * is not a ROUTER. |
| 19:07.34 | DelphiWorld | lol [TK]D-Fender sory! |
| 19:07.41 | DelphiWorld | [TK]D-Fender: but give me a pathern please |
| 19:07.50 | DelphiWorld | [TK]D-Fender: also if i don't say route, what i should say? |
| 19:08.01 | [TK]D-Fender | DelphiWorld: Call lands on an extension that matches the number |
| 19:08.26 | DelphiWorld | [TK]D-Fender: ok |
| 19:08.31 | [TK]D-Fender | DelphiWorld: Your dialplan calls applications. Which ones, and what they do is up to you and there is no "term" for this. |
| 19:08.51 | DelphiWorld | [TK]D-Fender: ok ok ok |
| 19:09.04 | DelphiWorld | want to dial 00calls but don't know how to |
| 19:09.47 | [TK]D-Fender | DelphiWorld: exten => _00.,1,NoOp(This matches any number starting with 00 that also has at least 1 more digit after the 00) |
| 19:09.58 | [TK]D-Fender | DelphiWorld: You don't seem to understand the dialplan at all. go read the book |
| 19:10.10 | [TK]D-Fender | DelphiWorld: And seriously, how long have you been coming here? |
| 19:10.55 | [TK]D-Fender | DelphiWorld: So far I see at least a year and a half. |
| 19:11.07 | DelphiWorld | [TK]D-Fender: ;) |
| 19:15.23 | DelphiWorld | [TK]D-Fender: i would by glad to see algeria tones added to dahdi;) |
| 19:17.06 | [TK]D-Fender | DelphiWorld: I'm sure they aren't that hard to write up. |
| 19:17.20 | DelphiWorld | [TK]D-Fender: ;) |
| 19:17.28 | DelphiWorld | [TK]D-Fender: if you need i can give you the tone spect |
| 19:19.13 | *** join/#asterisk riscphree (~riscphree@h92.9.91.75.dynamic.ip.windstream.net) |
| 19:19.48 | [TK]D-Fender | DelphiWorld: indications.conf <- go mod it up. I'm sure it isn't that complicated |
| 19:19.49 | *** join/#asterisk JParr (~JParr@24.244.133.116) |
| 19:20.23 | DelphiWorld | [TK]D-Fender: can i tel you what is my highest problem, |
| 19:20.33 | DelphiWorld | [TK]D-Fender: editing in unix like system. that's my problem |
| 19:21.29 | [TK]D-Fender | DelphiWorld: I fail to see how that is a "problem", and also don't have your other problems to compare against to validate that it's a real problem. |
| 19:21.50 | DelphiWorld | [TK]D-Fender: close your eyes |
| 19:22.01 | [TK]D-Fender | DelphiWorld: Open YOURS |
| 19:22.11 | DelphiWorld | [TK]D-Fender: if you can. |
| 19:22.29 | [TK]D-Fender | DelphiWorld: Trust me, the reverse is more likely challenging. |
| 19:22.58 | DelphiWorld | [TK]D-Fender: what do you mean by this? |
| 19:23.10 | [TK]D-Fender | DelphiWorld: Neverminid, I remember more about you now... |
| 19:23.22 | DelphiWorld | [TK]D-Fender: what do you remember... tel me? |
| 19:23.24 | [TK]D-Fender | DelphiWorld: Go use a GUI text editor then. |
| 19:23.34 | [TK]D-Fender | DelphiWorld: No-one said you had to use a TERMINAL based one |
| 19:23.55 | DelphiWorld | [TK]D-Fender: ok, i did it befaure you say it. i use WinSCP |
| 19:24.04 | DelphiWorld | but in this PBX is refusing to open a file |
| 19:24.28 | [TK]D-Fender | DelphiWorld: Its still just a text file. You also told me you're a FreeSWITCH user, so I guess the nauseating complexity of their XML isn't "too complicated" for you, so * sure as hell shouldn't be |
| 19:25.06 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 19:25.14 | DelphiWorld | yes, [TK]D-Fender i am very honest with this and don't want to start a discution that can't by terminated;) |
| 19:25.25 | [TK]D-Fender | DelphiWorld: I never said to use winscp. and if you screwed up your file permissions... I also don't know what you've done to create this problem. But that implies you don't understand hwo permissions even work. |
| 19:25.43 | [TK]D-Fender | DelphiWorld: And believe me, it CAN be terminated. NUMEROUS ways. |
| 19:25.47 | DelphiWorld | [TK]D-Fender: i am using exactly: switchfin |
| 19:25.55 | JParr | if i need to terminate a few PRIs to asterisk, is it preferable to use an external gateway, or interface cards in the server? |
| 19:26.07 | [TK]D-Fender | DelphiWorld: Anyway, go set up a Samba share to your * config folder and use your PC to edit the files. |
| 19:26.29 | [TK]D-Fender | JParr: Depends on what other rolse * would be doing for you, how many ports you need, redundency, etc |
| 19:26.33 | DelphiWorld | [TK]D-Fender: that's a good idea, but do switchfin runing in uClinux support it! DUDE |
| 19:26.39 | [TK]D-Fender | JParr: Describe the "big picture" |
| 19:27.02 | [TK]D-Fender | DelphiWorld: I can't think of a Unix-based OS that CAN'T run Samba. |
| 19:27.11 | JParr | [TK]D-Fender: calling cards for the moment, in the future, fixed line service over ip |
| 19:27.29 | DelphiWorld | JParr: good idea |
| 19:27.38 | [TK]D-Fender | DelphiWorld: And if YOU are complicating your physical situation by also picking troublesome hardware then you are asking for problems. |
| 19:27.59 | [TK]D-Fender | JParr: how many PRI's? |
| 19:28.09 | JParr | [TK]D-Fender: starting with 2 |
| 19:28.28 | [TK]D-Fender | JParr: And in the longer term? What is the rate of growth? |
| 19:28.53 | JParr | [TK]D-Fender: i couldn't say, but i doubt it'll ever grow beyond 8 T1s |
| 19:28.53 | [TK]D-Fender | JParr: What role will * server in the bigger picture? |
| 19:29.09 | [TK]D-Fender | JParr: Then I'd say still with internal cards. |
| 19:29.15 | [TK]D-Fender | stay* |
| 19:29.22 | JParr | ok |
| 19:37.43 | *** join/#asterisk chasing`Sol (~rc4@smtp.master-zone.net) |
| 19:45.23 | DelphiWorld | JParr: i suggest you to build asterisk based MGW's |
| 19:45.27 | DelphiWorld | mgw===media gateway |
| 19:45.29 | DelphiWorld | and trunk to it |
| 19:47.31 | *** join/#asterisk diegoviola (~diego@190.23.48.144) |
| 19:47.44 | *** part/#asterisk diegoviola (~diego@190.23.48.144) |
| 19:51.34 | *** join/#asterisk neurosys (~neurosys@c-65-34-190-58.hsd1.fl.comcast.net) |
| 19:53.15 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 19:53.24 | bluOxigen | hello all |
| 19:53.28 | bluOxigen | hi DelphiWorld :) |
| 19:53.57 | DelphiWorld | bluOxigen: PEEp |
| 19:54.05 | bluOxigen | [TK]D-Fender: hi, how you doing pal |
| 19:54.26 | bluOxigen | [TK]D-Fender: DelphiWorld is one of my old friends |
| 19:55.08 | [TK]D-Fender | bluOxigen: How many others does he have? |
| 19:55.22 | bluOxigen | :) |
| 19:55.37 | [TK]D-Fender | bluOxigen: Should we be expecting them here shortly? |
| 19:56.00 | DelphiWorld | [TK]D-Fender: why no;) |
| 19:56.01 | DelphiWorld | lol |
| 19:56.02 | [TK]D-Fender | bluOxigen: and should I be expecting a personal introduction from them as well? |
| 19:56.35 | DelphiWorld | [TK]D-Fender: only if mark spancer want! |
| 19:56.36 | DelphiWorld | :P |
| 19:57.03 | bluOxigen | [TK]D-Fender: no one is like him ...haha |
| 19:57.32 | bluOxigen | any ways, its not #delphiWorld its #asterisk :) |
| 19:58.11 | [TK]D-Fender | Delphi ..... Borland's attempt to keep Pascal relevant. |
| 19:58.27 | bluOxigen | ya.. Delphi was cool |
| 19:58.53 | [TK]D-Fender | bluOxigen: So were punch-cards in the day. Nostalgia isn't what it used to be. |
| 19:59.04 | bluOxigen | :) |
| 19:59.24 | bluOxigen | Delphi was way ahead of many other langs of that time, dont you agree |
| 20:00.13 | WIMPy | Wrong concept. We don't need good languages, we need good OSs again. |
| 20:00.33 | bluOxigen | infact if i m not mistaken MS hired the cheif architect of Delphi work on dotNet design |
| 20:01.24 | [TK]D-Fender | You say that .... as if it were a GOOD thing.... |
| 20:02.20 | bluOxigen | WIMPy: i think we all want some AI as good as it is shown in StarTrak |
| 20:03.41 | [TK]D-Fender | Artificial intelligence is no match for natural stupidity. Hang around here long enough and it will become quite evident |
| 20:12.27 | WIMPy | bluOxigen: Don't need that. Just a consitant design defined at OS level and not by used language or toolkit or whateva. |
| 20:13.53 | WIMPy | Th well known desktop environments are pretty unconvincing to the programmer and certainly to the user. |
| 20:14.51 | WIMPy | But luckily Asterisk is not a desktop application :-) |
| 20:20.05 | *** join/#asterisk Kyosh (whoa@pool-74-101-225-118.nycmny.fios.verizon.net) |
| 20:39.45 | DelphiWorld | to receyv a call through an FXO port, should i put my phone number as a pathern or what? |
| 20:41.42 | ChannelZ | no |
| 20:41.55 | ChannelZ | it'll go to the 's' extension of the context for that channel |
| 20:42.23 | bluOxigen | WIMPy: :) |
| 20:42.40 | DelphiWorld | ChannelZ: could you explin it ;) |
| 20:44.17 | ChannelZ | Hmm I think I just did |
| 20:45.06 | ChannelZ | your DAHDI channels have to have a default context. When a call comes in on one of your FXO channels, it will go to the 's' extension of that context in your extensions.conf |
| 20:47.33 | [TK]D-Fender | DelphiWorld: GO READ THE BOOK. |
| 20:47.35 | [TK]D-Fender | DelphiWorld: GO READ THE BOOK. |
| 20:47.37 | [TK]D-Fender | DelphiWorld: GO READ THE BOOK. |
| 20:47.45 | ChannelZ | do what now? |
| 20:48.53 | *** join/#asterisk ukine (~ukine@14-145.97-97.tampabay.res.rr.com) |
| 20:51.21 | *** join/#asterisk Ikarus (~ikarus@dhcp-077-250-195-114.chello.nl) |
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| 20:59.26 | Ikarus | Can I use the ALSA channel with multiple channels/alsa outputs ? or anyone know a commandline VoIP client other then linphone |
| 20:59.59 | [TK]D-Fender | Ikarus: No |
| 21:00.18 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 21:01.54 | *** join/#asterisk markit (~marco@88-149-177-66.staticnet.ngi.it) |
| 21:02.04 | markit | hi, any debian manteiner here? |
| 21:02.43 | Ikarus | [TK]D-Fender: meh :( |
| 21:05.24 | WIMPy | Ikarus: May the source be with you. |
| 21:18.40 | *** join/#asterisk Bryanstein (bryan@shellium/admin/bryanstein) |
| 21:19.55 | Ikarus | WIMPy: ... |
| 21:20.03 | Ikarus | WIMPy: need it working next thursday :) |
| 21:22.41 | WIMPy | Lots of time. |
| 21:25.15 | *** join/#asterisk DelphiWorld (~Delphi@41.200.30.82) |
| 21:25.25 | DelphiWorld | Ikarus: pjsua |
| 21:32.37 | Ikarus | DelphiWorld: isn't that OSS audio only ? |
| 21:33.20 | DelphiWorld | Ikarus: no, open source commandline softphone pjsua... |
| 21:40.18 | DelphiWorld | Ikarus: pjsua is pjsip user agent |
| 22:00.00 | *** join/#asterisk deonv (~adium@41.218.75.152) |
| 22:04.13 | *** join/#asterisk voxter (~voxter@76.77.73.130) |
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| 22:13.30 | DelphiWorld | night all |
| 22:19.34 | *** join/#asterisk dafresh (~dafresh@sge78-3-82-247-99-60.fbx.proxad.net) |
| 22:19.42 | dafresh | hi all |
| 22:20.40 | dafresh | i have a TC400 in my box, juste upgraded with ast 1.4.35 and dahdi 2.4.0, i am looking for cli cmd "show transcoder" |
| 22:21.13 | dafresh | card seems to be ok in dmesg with loading dahdi module, byt asterisk wont use it to do transcoding |
| 22:38.27 | *** join/#asterisk lanel (~chatzilla@2001:470:1f0b:1452:20a:cdff:fe10:c579) |
| 22:50.40 | ChannelZ | do you mean 'core show translation'? |
| 22:50.54 | ChannelZ | I'm not sure how/where the hardware transcoder shows up |
| 23:17.01 | *** join/#asterisk garymc (~chatzilla@host81-151-189-22.range81-151.btcentralplus.com) |
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