IRC log for #asterisk on 20070722

00:00.12Berny_getting the dial plan and all the regional settings 100% was a pest
00:00.58Berny_there was a couple of things it didn't like about my number conversions so I ended up with a dial plan a mile long, after all that though, it has been solid
00:01.08Berny_teething problems I wrote it downt o
00:01.10Berny_to eve
00:02.04*** join/#asterisk [hC] (n=hardcore@pool-71-112-142-43.sttlwa.dsl-w.verizon.net)
00:02.57x86perfect MoH material here
00:03.37Innatech_away"press one for WICKED TUNES."
00:04.02x86heh...
00:04.26x86customer calls for support, and while sitting in the queue they hear "HOLY SMOKE! HOLY SMOKE! THIS IS NO JOKE!"
00:04.30x86roflmao
00:04.57Berny_hehehe
00:06.45Berny_x86, just your beer comment, best place is belgium
00:07.10x86as long as you didn't try to say best place for beer is london ;)
00:07.11Berny_antwerp has a beer festival once a year, if you are around in europe, it's worth a look
00:07.25Berny_nup
00:07.25x86i'm a big fan of german beer, like Erdinger
00:07.35Berny_ah ok
00:07.39x86Honey Weiss
00:08.54*** join/#asterisk powerkill (n=powerkil@84.205.154.247)
00:09.01Berny_I think I had some of that in belgium
00:11.11Berny_anyway, I am outta here, thanks for the info x86. Just needed to know otherwise I would be running away with an idea before finding out if it's technically feasible or not
00:11.24*** part/#asterisk Berny_ (n=Berny_@ip-89-168-5-83.cust.homechoice.net)
00:36.45*** join/#asterisk Strom_M (n=strom@66.103.219.90)
00:54.42*** join/#asterisk zydrunas_ (n=zydrunas@24-119-29-130.cpe.cableone.net)
00:57.11*** join/#asterisk RazaMetaL (n=razameta@200.93.220.27)
00:57.14RazaMetaLhi guys
00:57.26RazaMetaLdoes any one using astribank ?
00:59.50*** join/#asterisk zydrunas_ (n=zydrunas@24-119-29-130.cpe.cableone.net)
01:03.55*** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca)
01:13.35*** join/#asterisk zydrunas_ (n=zydrunas@24-119-29-130.cpe.cableone.net)
01:22.15*** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca)
01:27.16*** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com)
01:34.37*** join/#asterisk Strom_M (n=strom@s142-179-221-179.ab.hsia.telus.net)
02:01.46*** join/#asterisk tako-san (n=Tako-san@154.5.212.245)
02:08.36MoutaPTx86 i'm not using ztdummy , i believe...
02:09.02MoutaPTi'm wondering if it is supposed to appear zap/pseudo on channels when using Meetme appliaction
02:33.07*** join/#asterisk Powerkill (n=Power@84.205.154.247)
02:50.01*** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com)
02:57.03*** join/#asterisk coppice (n=chatzill@93.195.17.210.dyn.pacific.net.hk)
03:03.51killfilli has my extension working in a phone. and now im unplug the phone, and try to configure form a software one, and  asterisk get: "Device does not match ACL".. i guess its saving the ip?
03:03.55killfillhow do i reset this?
03:05.00*** join/#asterisk grimbeans (n=audiokry@202.88.74.13)
03:05.09grimbeanshi fellas
03:05.29grimbeansI'm new to asterisk and have a question.. could someone please help me?
03:05.47grimbeansok here goes.. :p
03:05.57grimbeansI'm starting up a small internet telephony service locally..
03:06.04grimbeansI'm going to be using 16 analog trunk lines initially by way of analog FXO gateways
03:06.12grimbeansthe numbers will be 234-1001 through 234-1016
03:06.18grimbeansI want all my customers to call just 234-1001
03:06.23grimbeansas each call is received, I want them to be automatically transferred (call distributed?) to one of the other 15 lines
03:06.30grimbeansso that the original 234-1001 is constantly free and open to new calls
03:06.42grimbeansdoes AsteriskNOW's automatic call distribution feature allow me todo this?
03:06.56JTumm
03:07.08JTyou should NOT be using analogue lines to run an ITSP
03:07.14grimbeansi know.. T_T
03:07.28grimbeansbut the local telco provides T1 at $8,000 a month!!!!
03:07.33grimbeansdamn monopoly :(
03:07.50Hymiegrimbeans: you're paying for call transferring too?
03:07.52JTfor voice service?
03:08.03Hymieand, that would be cumbersome
03:08.26grimbeansthe local telco apparently does not provide call forwarding on busy
03:08.30Hymiewhy not just pay for a ring down service from the telco
03:08.39grimbeansand i think they don't distinguish between voice/data T1 here
03:08.44grimbeansa ring down service?
03:08.47Hymieyes
03:08.51Hymieit's a fairly standard thing
03:08.52grimbeanswhat is that? (i'm noob) :(
03:08.54Hymieyou dial one number
03:09.09Hymieand the telco sets it up so it will find any free line and ring that, in the ringdown group
03:09.23Hymiethat's how most companies handle having 5 or 10 lines
03:09.35grimbeansthat would be great :D
03:09.52grimbeansI just hope my local telco is willing to provide that service..
03:09.52coppiceits more commonly called a hunting group
03:09.55HymieI'm not sure if there is an upper limit on the number of ring down lines allowed
03:10.17Hymiecoppice: probably I'm speaking local telco lingo.. they call it ring down lines here :Þ
03:10.26grimbeansic ic.. :)
03:10.29*** join/#asterisk newsmafia (n=newsmafi@wsip-68-224-174-204.sd.sd.cox.net)
03:10.39grimbeansI guess I ought to call the telco on Monday to get answers..
03:10.47Hymieindeed
03:10.51Hymieyou pay for the service though
03:10.59Hymienot sure what the extra $$$ is
03:11.06Hymieof course
03:11.13grimbeanshopefully it's not too crazy a price :p
03:11.20Hymieare you planning to use voip for the dialout service?
03:11.25Hymieis this part of the asterisk plan?
03:11.25grimbeansyep :)
03:11.32Hymiethen why not use voip for incoming?
03:12.07Hymieit's probably cheaper, althugh you'll need more internet bandwidth
03:12.25grimbeanshmm..
03:12.45Hymiethat's your biggie there,b tw... because if you think you're going to run this service on a cable modem or dsl, just walk away right now
03:13.07Hymieyou're going to need fibre or host this at a local isp with a GOOD backbone
03:13.14Hymieotherwise, your customers will walk
03:13.21grimbeansreliability issues huh :(
03:13.34Hymiewell sure, if your bandwdith / isp is crap, that's not going to work well
03:13.49Hymiewhat did you plan for your bandwidth?
03:14.04grimbeanswell, in the beginning I was going to start with just a 1.8 mbps cable line
03:14.15Hymiewhat's the upstream on that?
03:14.16grimbeansand add more cable bandwidth as necessary
03:14.34grimbeanshmm must be like 1/3 of it or so.. I gotta check.. :(
03:14.48grimbeansyou're right, it's the upstream that matters :(
03:15.06Hymiedoesn't sound to good to me.. I mean, that's probably residential, yes?  even if not, cable modems and cable ISPs aren't known for their stellar backbones/routing
03:15.35grimbeanswell, it's a "commercial" plan.. and the highest available here :(
03:15.38Hymieyou might do better to find a GOOD isp, host a box there, and use a GOOD voip provider for incoming and outgoing
03:16.05Hymiethe ISP will likely have free UPS service, as well as a generator with days of gas/diesel
03:16.34Hymieif you factor in that, your electricity costs, the commercial cable modem costs, and the costs of the telco lines, >I think the ISP might actually be quite cheaper
03:16.38Hymieas long as you get a *good* isp
03:16.41Hymiewith a good backbone
03:16.47coppiceUncontrolled Power Supply :-)
03:16.52Hymieheh
03:17.32Hymiegrimbeans: with the money you save on installing those 15 phone lines, you could buy a second rack mount machine, and set it up using heartbeat or soemthing to fail over in case the first fails
03:18.15killfillhm.. i have a queue defines with 2 agents. When this 2 agents are talking (i.e. busy), when making a third call to the queue, this new call never got sent to any agents. i wish that the agents phones blinks the second line led or something.
03:19.02Hymieyou could do that with polycoms.. what phones do you have?
03:19.14Hymieshould be easy to do with any phone that has multiple presences
03:19.41killfillgrandstream 2000
03:19.59killfillit works when you call the phone directly (i.e. the sip extension) but not for the queue.. :S
03:20.20killfillhttp://pastebin.ca/629288 <-- thats my queue
03:20.30Hymieif you have multiple sip logins for those phones, you could just pretend they are multiple phones
03:20.34Hymieif you see what I mean
03:21.13killfillyes.. but i wish not to do that
03:21.43killfilli would need to double the config (users.conf), and every user will need to login/logout as agent several times per phone :)
03:21.44Hymiethen I know not what to say... I haven't played with queues very often.. but from what you say it sounds like there is no call waiting on them(whicvh makes sense, it's a queue after all..)
03:22.44killfillhm,..
03:23.09Hymiekillfill: this place is dead right now, it's usually hopping during the week
03:23.18Hymiekillfill: if you're looking for more queue info, or more certainty on queues
03:23.26Hymiekillfill: I'd try then...
03:24.07killfilltry what?..
03:24.14Hymieyour questions
03:24.32killfillah.. try then in the week.. :P
03:24.37killfillok..
03:24.41Hymieyeah
03:24.50HymieI thinkmost people here are here on their work machines
03:24.58grimbeanshymie, thanks for all the advice :) :* but I think there are some limitations to my local area that make a lot of the options you suggested impossible for me..
03:25.11Hymiegrimbeans: such as?
03:25.15grimbeanslet me explain my situation in a bit more detail.. :)
03:25.18grimbeansfor one,
03:25.39*** join/#asterisk |dennis| (n=dennis@200.32.236.20)
03:25.44grimbeansI'm on an island in the pacific ocean, and the only close area is Guam
03:25.53grimbeansthere are only 2 isps
03:25.58grimbeansthe telco, and the cable company
03:26.19Hymieok, your isp can be anywhere in the world
03:26.21Hymiekeep that in mind
03:26.23grimbeansand I will be the only ITSP on this island when I start in the weeks to come
03:26.24grimbeanshmm
03:26.44Hymieso, what's more important is if someone already offers DNDs on your island
03:26.55Hymiecheck places like les.net to see ift hey do offer phone numbers there
03:27.00Hymieif they do, you can use them for incoming and outgoing
03:27.09grimbeansdnds? are they like dids?
03:27.17Hymiewhat country is this island?
03:27.20Hymiesure, dids I mean
03:27.30Hymiealso, for example, les.net has 800 numbers
03:27.40Hymiewhich might be an option if you are outside of local did ranges
03:27.47grimbeansthis is Saipan, of the Northern Mariana Islands, which is a protectorate of the U.S. (has a U.S. zip code, area code etc.)
03:27.52grimbeanshmm
03:28.01Hymiecan you call US 800 numbers?
03:28.14*** join/#asterisk bkruse_home (n=kruz@69.73.127.92)
03:28.30Hymiedo you exist within the confines of thge US area code system?
03:28.32grimbeanssome 800 numbers work, but more often not.. the only way for me to find out is to specifically call each 800 number from here to see if they work
03:28.42grimbeansyes it has area code 670 :)
03:29.02*** join/#asterisk vitaminmoo (n=vitaminm@70.58.177.109)
03:29.48grimbeanshmm so are you telling me that if I could access les.net (for example) by 1-800 access, I could use their high speed internet service?
03:30.12Hymieno, you'd use their 800 did for incoming calls, then go out via their voip service
03:30.18Hymiefrom an ISP you setup on domestic US soil
03:30.25Hymiea big, reliable, cheap one
03:30.43grimbeansoh, ok.. I see..
03:30.45grimbeanshmm
03:31.12Hymiesec
03:31.24grimbeansok :)
03:31.54Hymiehmm
03:31.54Hymieyeah
03:31.57grimbeansokay I've actually thought of that earlier when I was brainstorming for options
03:32.01Hymieyou'd have to do 800 for les.net it looks like
03:32.13Hymiethat's your biggie, is 800 supported in your area
03:32.17grimbeansI opted not to go with it for a few reasons, although it would have been more stable
03:32.19Hymiebeucase, this is waaaay cheaper for you
03:32.23grimbeanshmm
03:32.32Hymieit scales better too
03:32.40Hymiedynamically
03:32.49Hymieyou don't have idle phone lines, or too few phone lines
03:32.54Hymieyou have the precise right number of phone lines
03:34.39grimbeansyeah i'm here :)
03:34.45grimbeansi'm just thinking through what you've said
03:35.05Hymiethat's a les.net 800 number
03:35.12grimbeansok i'll try now :)
03:35.33_DAWI dont think you will find many ITSP's selling dids in the NM islands :)
03:35.44Hymie_DAW: there are thusands of them!
03:35.55_DAWreally..  who?
03:35.59Hymiewhy
03:36.01Hymiethat guy you know
03:36.02Hymieand
03:36.07_DAWjust curious
03:36.08Hymiethe other dude that hangs out with buddy
03:36.09Hymieyou know
03:36.12Hymiethe one with the face
03:36.37_DAWvery nice..  I see you have it all lined up.
03:36.51Hymiehehe :Þ  not sure about 800 number penetration though
03:37.12grimbeansI tried calling the number and it seems like it works..
03:37.17grimbeansit gives me a short beep
03:37.30Hymieuh
03:37.36Hymieit didn't make it through to me ;)
03:37.43grimbeanswhich is unlike the recorded "this toll free number is not accessible from your service area" thingie
03:37.44Hymieso, it probably doesn't work :(
03:37.45grimbeanshmm
03:37.48grimbeansdarn
03:37.50grimbeans:p
03:37.51grimbeansoh well
03:38.07grimbeansjust confirmation of the
03:38.17Hymiebut, emnail the les.net dudes!
03:38.27Hymietell them what you want
03:38.31grimbeanspoor connection that nmi have to the world :p :)
03:38.32Hymiethey may say tough
03:38.34Hymiebut you'll know
03:38.36Hymiealso
03:38.43Hymiethey do answer the phone almost immediately
03:38.48Hymieduring normal hours
03:38.52Hymieso, you can try that too
03:39.10grimbeansok I'll try it when it's work hours in the U.S. :)
03:39.42grimbeansnow let me explain a bit about why I was (and so far, still am) planning to do this the manual, cumbersome way, despite the instability
03:40.23grimbeanshmm.. maybe I need to think it through again lol..
03:40.28grimbeansbut here's what I thought
03:41.01grimbeansinstead of the expensive T1 at $8,000 and only 24 voice lines,
03:41.09*** join/#asterisk MACscr (n=MACscr@adsl-75-23-73-117.dsl.peoril.sbcglobal.net)
03:41.13grimbeansI would use regular analog lines at about $50 each
03:41.32grimbeanswhich comes out to $1,200 for 24 lines / mo
03:41.34Qwellwho charged $8k for a T1?  That's pretty ridiculous
03:41.48Hymieit's cheaper than your crazy telco charges for that t1
03:41.48Hymiebut
03:41.54MACscrDoes anyone know of a outbound provider that allows for complete callerid rewriting? This includes the CNAM
03:41.54grimbeansyeah, it's RIDICULOUS but my telco has a monopoly on a lone island and there's no alternative
03:41.56Hymiethere is also another issue ; installation coats
03:41.58coppiceyou should try to get that in the Guiness Book of Records - "Most expensive T1 in history"
03:42.01Hymieif you go to put in all those lines
03:42.08Hymiethey're going to carge you big tiem for installation
03:42.27grimbeans:)
03:42.28grimbeanshmm
03:42.32grimbeansabout the installation costs
03:42.42grimbeansI'm setting up shop in a major building
03:42.44Hymieno,you can't borrow barbra eden
03:42.47grimbeansthat has
03:42.58grimbeanslike over a hundred units
03:43.04Hymiegood
03:43.15Hymieso you only have to write from the phone close to your server room
03:43.18Hymiewire
03:43.26grimbeansyep :)
03:43.29HymieI think you'll be sorry with your cable though
03:43.31Hymiefor bandwidth
03:43.37grimbeansT___T
03:43.41grimbeansit's either the cable co
03:43.45grimbeansor the telco
03:43.53HymieI bet their cable isn't separate from their residential cable
03:43.58Hymiesame for the telco
03:43.58grimbeansand the telco will hate me for taking their revenue and they suck anyways :(
03:44.04Hymieprobably uses the same backbone, etc
03:44.09grimbeansyeah it's all together
03:44.17grimbeanssame backbone :(
03:44.25Hymieif you can, you should *really* get into an ISP on the continent
03:44.27Hymiea good one
03:44.41Hymieif you can somehow get dids or 800s working locally
03:44.45grimbeansI guess I should think about that more seriously..
03:45.17grimbeansI've considred DIDs but I don't think anyone is providing DID service to here
03:45.32grimbeanswhich seems logical since there is no ITSP here (except me starting up :P)
03:45.38grimbeansand as for the 800 service
03:45.42grimbeanssome services may work
03:45.44*** join/#asterisk andresmujica (n=andresmu@201.245.236.215)
03:45.54grimbeansbut I think they charge me like $.02 per minute
03:46.05grimbeanswhich increases the rates I can offer to my customers :(
03:46.19Hymiewell, you only need to undercut the telco
03:46.20grimbeanshmm
03:46.29Hymieand, international rates usually shine with voip
03:47.06grimbeansyou're right.. :)
03:47.22grimbeansI probably should re-calculate costs before deciding
03:47.25grimbeansand
03:47.34grimbeansi should also call the telco on monday to check on the
03:47.37coppicewe can't afford to use VoIP for international calls. We have to use the local telcos to save money :-)
03:47.50grimbeansringdown service
03:48.23grimbeansbut back to square one..
03:48.41*** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn)
03:48.41grimbeansif the 800s don't work out and if the telco refuses to provide ringdown service
03:48.57grimbeansI may need to rely on Automatic Call Distribution
03:49.02grimbeansto make it work at all
03:49.16grimbeanswould it even be possible using Asterisk?
03:49.47*** join/#asterisk tuxd00d (n=tuxinato@128.187.178.113)
03:49.50HymieI've neerevern heard of that
03:50.04grimbeansT__T
03:50.08grimbeanswell I've been reading for days
03:50.10HymieI really think you're barking up the wrong trr
03:50.14grimbeansT__T
03:50.17Hymiehow is it going to move the call to another line?
03:50.20Hymiecall transfer?
03:50.30Hymieit has to tell the telco to move the call, after all
03:50.39grimbeanswell that's what I thought
03:50.53grimbeansuntil I searched and searched and got some information about
03:50.57grimbeansACD
03:51.05shido6?
03:51.17grimbeansand now I'm confused just as to what is possible with ACD
03:51.33shido61 # ?
03:51.36grimbeansI'm wondering if it can only distribute WITHIN the network or IP
03:51.38shido61 channel?
03:51.55grimbeansdo you mean 1 analog channel? (i'm noob sorry :()
03:52.09grimbeansi plan to use 16 analog trunk lines initially
03:52.12shido6is the 8xx number coming in voip or pri?
03:52.15shido6or pots
03:52.34*** join/#asterisk bmg505 (n=leon@196.209.178.115)
03:52.37grimbeansit's just a local access number with all regular analog POTS lines (no T1, etc.)
03:52.45grimbeans(T1 costs $8,000 here :()
03:52.47shido6how many pots lines?
03:52.49shido6good lord
03:52.53grimbeanshaha yeah
03:52.53shido6where are you ? Moon ?
03:53.00Qwellshido6: Saturn, I think
03:53.03grimbeansmiddle of the pacific ocean T________T
03:53.04shido6in a volcano?
03:53.16grimbeansnow it's a lifted limestone island but
03:53.21shido6no SAT?
03:53.25grimbeansneighboring islands are volcanoes! that erupt!
03:53.25shido6VSAT
03:53.26shido6?
03:53.31grimbeansVSAT? satellite?
03:53.40shido6yeah
03:54.07grimbeansI'm not so familiar with satellite service (except satellite TV)
03:54.16shido6google it
03:54.18grimbeanswhat is possible? there are some satellites that can be tapped here
03:54.19shido6save your $8k
03:54.22grimbeanswell
03:54.26grimbeanshaha
03:54.27shido6250 ms
03:54.42grimbeansthe thing is I need
03:54.46*** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca)
03:54.51grimbeansto let people from my local area call me up
03:54.57grimbeansusing PSTN
03:54.59coppiceyour calls go sailing on the Latent Sea
03:55.16grimbeans250 ms really good for my location XD
03:55.34grimbeansit's like 200 ms - 500+ ms usually lol
03:55.41grimbeanseven on wired broadband
03:55.52shido6... yeah
03:56.00Qwellgrimbeans: what island?
03:56.11grimbeanssaipan, of the northern mariana islands.. near guam :p
03:56.37grimbeanshmm
03:57.05shido6tap into the fiber from the Commonwealth of Northern Mariana Islands Public School System
03:57.08grimbeansyeah anyways T1's $8K so i'm planning to just use regular analog lines, $50 each in conjunction with cable internet
03:57.09shido6they will never know...
03:57.11grimbeanshmm
03:57.12shido6(gulp)
03:57.13grimbeanshahaha
03:57.45shido6actually
03:57.49shido6you should ask.
03:57.56JTgrimbeans: you should probably just give up
03:58.02QwellJT++
03:58.03grimbeansnooo T_T
03:58.22grimbeanswell
03:58.24grimbeansultimately
03:58.25shido6they have a T1 to the internet
03:58.34grimbeansthere's only one fiber optic cable
03:58.39grimbeansthat connects this island to the world
03:58.49shido6oh wow
03:58.49grimbeansand that cable is owned by the local telco
03:58.52grimbeansand
03:58.54shido6how many channels did you need?
03:58.56grimbeanseveryone else just rents it
03:58.58shido6(simultaneous calls)
03:59.10grimbeansat first just 16 to keep costs low but
03:59.21grimbeansi expect I'd need like 50 to
03:59.25shido6yeah
03:59.28shido6thats gonna suck
03:59.33grimbeanspossibly a few hundred eventually
03:59.47QwellYou're never going to get a few hundred analog lines...
04:00.01shido6is there a GSM carrier over there?
04:00.22grimbeansthe local telco just recently (a week or two ago) got GSM
04:00.29grimbeansnobody else has it :(
04:00.49grimbeansqwell, is that because of the installation costs?
04:00.52shido6is the local telco an enemy today?
04:01.02QwellNo, they just aren't ever going to give you that many
04:01.09Qwellnot when they're charging $8k for a T1
04:01.11grimbeansnot yet, but it will be very soon once I take their long distance and some local customers away
04:01.16grimbeanswell
04:01.20grimbeansthat's one thing though
04:01.26grimbeansthey have a monopoly
04:01.32grimbeansthey can't refuse service to anyone :D
04:01.40Qwellsays who? :p
04:01.51grimbeansmy lawyer anthony long that's who! :D
04:01.58grimbeanshehe
04:02.02grimbeansand in the mid '90s
04:02.11shido6you're going to need "two-way" satelite internet
04:02.15grimbeansthe telco got sued for antitrust violations once
04:02.18grimbeanshmm
04:02.25grimbeansshido,
04:02.37QwellIt would probably be cheaper to run a second fiber link to the island :p
04:02.41grimbeanswould the satellite internet be any better than the cable internet i'd be getting?
04:02.46grimbeansexcept maybe stability
04:02.50grimbeanshaha another fiber XD
04:02.57coppiceSaipan, land of a million cheap tee-shirts :-)
04:03.05grimbeansit cost the telco like 16 million $ to connect saipan to Guam :D
04:03.10coppiceThis is the main fibre activity
04:03.46shido6how many ppl will you service?
04:03.59sevardlulz, how many people will you service.
04:04.01sevardyour mother.
04:04.10grimbeanswell, the last company that tried something like this (and went under)
04:04.26shido6fill out the form and call http://www.macrosat.com/macrosat-reseller.html
04:04.27grimbeanswas using that $8,000 T1 line and they lasted maybe a bit short of a year
04:04.32Qwellgee, I wonder why they went under...
04:04.42shido6bbl
04:04.56grimbeansactually it wasn't a revenue issue but some of the managers were diverting money.. :(
04:05.30grimbeansmacrosat? is that the satellite internet service provider?
04:05.36Hymieat $300 a line it wasn't a revenue issue?!
04:05.38JTshido6: forget sat
04:05.42JTshido6: waste of time
04:06.18coppicesatellite is the only way he will bypass the telco.
04:06.25grimbeansyeah they were doing quite well with the long distance calling service
04:06.33Hymiehe could use a blimp ;)
04:06.34grimbeansoften all lines would be busy
04:06.42JTbypassing the telco for resale would appear to be a waste of time
04:06.43grimbeanshaha
04:06.49grimbeansi think
04:06.56grimbeansi gotta use the cable company for now
04:07.04grimbeansbad as that sounds :(
04:07.07HymieFIND OUT ABOUT THE 800 NUMBERS
04:07.07QwellJT: s/^.*\(resale\).*$/\1/
04:07.11JTor just not do it
04:07.12grimbeansyup
04:07.13grimbeansi will
04:07.15Qwellerm
04:07.18*** part/#asterisk andresmujica (n=andresmu@201.245.236.215)
04:07.19grimbeansi'll check the 800s
04:07.20QwellJT: s/^.*\(resale.*\)$/\1/
04:07.43grimbeansyou mean resell voip service? :p
04:07.57JTi mean forget about it, seriously
04:08.02JTunless you have big dollars
04:08.16grimbeanswell
04:08.21HymieJT: he does, he used a photocopier to enlarge some ones
04:08.39grimbeansif i go the analog route.. here's what it would cost me initally with 16 lines
04:08.43grimbeansper month
04:08.46grimbeansi'd have to pay
04:08.52grimbeans$1,200 for pstn
04:08.54*** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com)
04:08.56JTdoes the telco run the cable?
04:08.57grimbeans$110 for cable
04:09.18grimbeans(they own the fiber optic cable but the cable company rents it and telco can't stop them)
04:09.25grimbeans$1,000 for rent
04:09.28JTwhat's the upstream on the cable?
04:09.31grimbeans$400 for utilities
04:09.39grimbeansI'm guessing 768 kbps :(
04:09.42grimbeansI may have to add
04:09.46grimbeansanother line or two of cable
04:09.53grimbeansbut that's pretty cheap so it's ok
04:10.02grimbeansand
04:10.07killfillQwell: may i ask you if its possible to make a queue act as callwaiting?  i.e. when all agents are talking, and a new calll cames in, i wish that that call comes into the queue, and make the agent's phone blink.
04:10.32JTgrimbeans: what is the primary type of service you are offering? incoming DIDs, or outgoing calls?
04:10.37killfilleventually they could anwear the new call, putting the current one in hold.
04:10.46grimbeanscall waiting service is available from the telco.. call forwarding on busy is not provided
04:10.46coppicegrimbeans: you seem to be treating the internet bandwidth like its guaranteed.
04:10.54grimbeansand jt i'll be giving out
04:10.57grimbeansoops
04:10.58killfill(or anyone..)
04:11.05JTcall waiting is completely useless to a pbx
04:11.05grimbeansi'll be providing three things
04:11.58grimbeans1.) voip service to those who have broadband, 2.) prepaid long distance calling service, and 3.) local DID (considering cost currently)
04:12.18JTlocal did is the hardest
04:12.26grimbeansyeah i know :(
04:12.28JTwithout pri, it really is a waste of time
04:12.31JTor ss7
04:12.34grimbeansT_T
04:12.47grimbeansi may have to just stick with the first two if the DID is not viable..
04:12.54Qwell~wglwat
04:12.55jbotwglwat is, like, well, good luck with all that
04:13.00bkruse_homeQwell++
04:13.08wunderkin-wigglewat
04:13.10grimbeanshehe. thanks guys :*
04:13.12Qwellbkruse_home: wanna buy a T1 from Qwellcomm?
04:13.21bkruse_homeQwell: ill take 2
04:13.21Qwell$7500 - that's a real bargain
04:13.26wunderkin-telecomjoshvoxmart?
04:13.40bkruse_homeqwellcomm bought telecomjoshvoxmart out
04:13.41Qwellwunderkin-: Qwellcomm bought telcomjoshvoxmart
04:13.50bkruse_homethats old news.
04:13.53wunderkin-i missed that, will be at&t before you know it
04:13.58grimbeansxD
04:14.07bkruse_homewe are working on it currently.
04:14.18QwellQwellcomm, the new AT&T
04:14.28wunderkin-sounds close enough to qualcomm
04:14.30bkruse_homei like it
04:14.33Qwellwunderkin-: yes
04:14.50Qwelland I'm gonna get sued for it one day, I'm well aware of that.
04:14.56grimbeans:D
04:15.28grimbeans(sorry to sound like a broken tape recorder but)
04:15.40bkruse_home-v grimbeans :P
04:15.42grimbeansso asterisk would not be able to let me do the following?
04:16.00grimbeansI'm going to be using 16 analog trunk lines initially by way of analog FXO gateways
04:16.06grimbeansthe numbers will be 234-1001 through 234-1016
04:16.10grimbeansI want all my customers to call just 234-1001
04:16.17JTwhat sort of fxo gateways?
04:16.17grimbeansas each call is received, I want them to be automatically transferred (call distributed?) to one of the other 15 lines
04:16.23grimbeansso that the original 234-1001 is constantly free and open to new calls
04:16.26grimbeanshmm
04:16.32grimbeansthey are audiocodes mediapack 108
04:16.44grimbeansx 2 or x 3 as needed
04:16.46JTalso, if you telco cannot do disconnect supervision you are also wasting your time
04:17.10grimbeansdisconnect supervision? i'm not familar with it .. help :(
04:17.24JTpolarity reverse on far end disconnect
04:17.30JTonly needed for analogue lines
04:17.35JTbecause they have crap signalling
04:17.47JTso a computer/pbx can automatically detect the other end has disconnected
04:17.58JTotherwise you could be left with zombie lines for periods of time
04:18.36grimbeanshmm.. so to find out, I should ask the telco whether they have disconnect supervision capabilities?
04:18.39HymieJT: problem is, the lad only has one telco... so... other than going 100% voip, and setting his box up via some isp in the US.. and using an 800 did....
04:18.57HymieJT: he has no options.. so, even if his local phone line is a set of cans and a string, the fxo module better handle that ;)
04:19.42JTHymie: i understand, sometimes the best option is to give up while you're ahead
04:19.50grimbeansnoo XD
04:19.51JTgrimbeans: yes
04:19.55JTfind out
04:19.55grimbeansgyaa
04:20.14grimbeansok i'll check on monday (their office is closed atm)
04:20.16grimbeans:(
04:20.22grimbeansso
04:20.37grimbeansif they have disconnect supervision capability,
04:20.53grimbeansasterisk would be able to do the automatic call distribution?
04:21.02grimbeansvia my fxo gateway
04:21.11JTi have no idea what you mean via automatic call distribution
04:21.24grimbeanswell supposedly
04:21.39grimbeansit can do all that call transferring / routing that I need
04:21.48grimbeansbut I just am not sure I'm undestanding it right
04:21.53grimbeansfor example
04:21.55grimbeanscall centers
04:21.58JTi'm not sure what you mean
04:22.10grimbeansuse Automated Call Distribution (ACD) systems to
04:22.13*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
04:22.29grimbeansreceive calls on one number and route them to various agents in their office
04:22.29JTdoesn't that have more to do with agents?
04:22.33grimbeansyeah
04:22.41grimbeansbut it can apparently also be used for voip
04:22.46grimbeansso basically
04:22.48JTnot much to do with being an ITSP
04:22.51grimbeansmy customer calls 234-1001
04:23.01grimbeansand it automatically routes to some other of my 15 lines
04:23.08grimbeansand frees up 234-1001 for more incoming calls
04:23.26JTwhen your customer calls a number, your telco will put the call through to a line
04:23.37grimbeanswell it's related to ITSP in the sense that
04:23.47JTif you have line hunt, it may put it through to the next available line in a group of lines
04:23.50grimbeansprepaid long distance calling will be accessible to customers by calling my number 234-1001
04:23.53grimbeanshmm
04:24.14grimbeansline hunt / ringdown, as hymie and you mentioned earlier huh..
04:24.20grimbeansanother thing I should check with the telco..
04:24.32JTand if you want to do analogue DIDs, you either need seperate lines/numbers, distinctive ring, or dtmf upon connect, all which are dodgy and not scalable
04:24.38grimbeanslast time, I only asked about call forwarding on busy signal and they didn't have it
04:24.42grimbeanshmm
04:24.48JTforget large scale DIDs on analogue
04:24.52JTutter waste of time
04:24.55grimbeansok :(
04:24.58grimbeansno DIDs then :(
04:25.03JTi mean
04:25.11JTyou could have numbers for pre-paid calling
04:25.12grimbeansonly voip service and prepaid long distance :(
04:25.18JTjust can't have hundreds of numbers
04:25.36grimbeanshmm ok let me digest the above info .. :p :)
04:26.47grimbeansand
04:27.02grimbeansthe reason why i can't have hundreds of numbers is..
04:27.08JTyou'll hate me, but where i am, i can get a 10 channel pri for about USD$175/mo :P
04:27.08grimbeansbecause the telco won't give it?
04:27.18grimbeansi hate you :( DIE!
04:27.22grimbeanslol
04:27.32wunderkin-yeah,me too
04:27.35JTbecause you have insufficient signalling to determine what number is being callED
04:27.45grimbeansbut
04:27.56grimbeanswouldn't callerID do the trick?
04:28.03grimbeans(that's what i was counting on)
04:28.23JTthat tells you who is callING
04:28.32JTunless your telco is willing to change that
04:28.34[TK]D-Fendergrimbeans, You are dangerously clueless.  Get a good lawyer
04:28.43JTto who is being callED, i doubt that they will
04:28.45grimbeanssorry TK i'm noob T_T
04:28.53[TK]D-Fendergrimbeans, And some industrial strength antacid
04:29.09grimbeans:p
04:31.08grimbeansThanks for all the advice, guys.  I really appreciate it. :)
04:42.51[TK]D-FenderCrushing telco-wannabe's one newb at a time.....
04:43.15coppice[TK]D-Fender: you mean a dead lawyer? aren't they the only good kind?
04:44.05[TK]D-Fendercoppice, no, living ones make better meat-shields when the ninja's come for you ;)
04:44.12*** join/#asterisk gardo (n=gardo@121.97.253.3)
04:44.29coppiceI guess everything has some useful application
04:47.28*** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca)
04:49.24*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
04:49.39*** join/#asterisk evilicey (i=jovens@cthulhu.got-unix.net)
04:53.53eviliceyAnyone have any recommendations for a softphone ?
04:56.30CunningPikeevilicey: SJPhone
04:57.23Strom_MCunningPike: i have a canada question for you :)
04:57.45Strom_Mdoes there exist in calgary such a thing as a 24 hour drugstore, and if so, what would it be called? :)
04:58.06coppicea pusher? :-\
04:59.40*** join/#asterisk |dennis| (n=dennis@200.32.236.20)
05:01.14eviliceythanks, i will give it a try :)
05:01.19eviliceyalso looking at X-lite
05:04.15JuggieStrom_M, you there still?
05:04.26Strom_Myes?
05:04.31Juggieshoppers drug mart
05:04.40Juggiewill have a 24hr location somewhere
05:04.45Strom_Myes, i just let my fingers do the walking (tm) and discovered this
05:05.00Juggieyour visiting calgary?
05:05.16Strom_Myes
05:05.23Strom_Ms/your/you're/
05:05.49Strom_Mare you in 403 land?
05:06.42Juggieno
05:06.44Juggiei'm in ottawa
05:06.59Juggiealso if you want to see if you can find something closer, you can also try rexall
05:07.02Juggiewww.rexall.ca
05:07.19Juggiei just know the two big drug store chains are shoppers & rexall :)
05:07.20*** join/#asterisk red9012 (n=marc3234@206-248-161-74.dsl.teksavvy.com)
05:07.58*** join/#asterisk shinao1 (n=shinao1@41.205.188.93)
05:08.32red9012how do online fax->email services work. which software are they using?
05:09.31JTthere are probably many different setups
05:10.09Strom_MJuggie: yeah...shoppers drug mart is the only one which seems to go to the trouble of actually pointing out the 24 hour bit
05:10.37Juggiestrom, yeah i just quickly looked @ rexall out of curiosity and it seems they are all closed by 10.
05:10.58Juggiethe shoppers by me are all open until 12 midnight or later hence why i suggested
05:11.35Juggieit should only be like 11:15 in calgary now anyways so should be plenty open
05:15.14Juggiehope you find what you need :)
05:15.18killfillhm..
05:28.12*** join/#asterisk jicksta_ (n=jicksta@S01060080c828ac2f.vn.shawcable.net)
05:36.35JTall you north amercans
05:36.38JTamericans
05:36.49JTif someone from overseas calls you, do you have to pay?
05:36.56JTif so, is it higher rates than normal?
05:37.28grimbeansno (as long as it's a landline), and no.. :p
05:37.39grimbeanserm you do mean PSTN right?
05:37.57JTyes
05:38.07grimbeansthen no, and no.. :p
05:38.15grimbeansonly they have to pay not you
05:38.26JTwell in the US, if you're not on an unlimited plan, i know you need to pay to receive calls usually
05:39.37grimbeanshmm.. I'd personally never noticed charges for calls received, on the telco's phone bill, except for collect calls of course
05:39.58grimbeansthey didn't even charge local calling rates when I received.. but that could just be a local thing where I was back then
05:40.01DarKnesS_WolFwhat is the best billing system for asterisk ? " webinterface "?
05:40.23JackEStormon landlines no, unless you have a grandparents plan, or a DOD.
05:41.27JTJackEStorm: but on mobiles you do?
05:42.29grimbeansfor mobiles it probably varies from region to region, provider to provider
05:42.50grimbeanson my little island they charge all calls received on mobile as though I were calling
05:43.05grimbeansbut when I was back in the mainland they didn't charge for calls received on my mobile
05:43.20grimbeansso it probably varies
05:43.26JTheh, billing is such an utter mess over that way :P
05:44.05grimbeans:)
05:44.48JuggieJT, incomming calls are allways free.
05:45.00JTon mobiles apparently not
05:45.07Juggieexcept mobiles yes
05:45.10Juggielandlines are allways free
05:45.28Juggiebut the rates dont change
05:45.34Juggiefor mobile, they just use your minutes
05:46.09Juggieunless of course you are roaming
05:46.11Juggiebut thats different
05:47.59JThmm ok
05:50.10Juggieanyways, it makes no difference where the call orginated from
05:50.18Juggieyou get charged the same
05:50.59JTah ok
05:51.11JTthe person i want to call is on nextel on generally only uses it for ptt
05:51.19JThe says his call rate is super high
05:51.42Juggiethats his plan though, not where the call comes from
05:51.49JTright
05:52.07Juggieany provider should offer unlimited evenings & weekends for cheap
05:52.29JackEStorm'cept on burners
05:52.42Juggieburners?
05:53.00JackEStormprepaid phones
05:53.38CunningPikeStrom_M: Sorry - I was away for a bit
05:54.53JuggieJackEStorm, i guess.... plans are cheap
05:55.05Juggiei pay something retardely cheap, like 20$ + SAF
06:37.06*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
06:43.37*** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca)
06:45.00*** join/#asterisk menil (n=meni@bzq-179-153-130.static.bezeqint.net)
06:53.05*** join/#asterisk jer (n=jtregunn@unaffiliated/jer)
06:57.05*** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com)
07:29.15*** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com)
07:44.06*** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca)
07:55.42*** join/#asterisk |dennis| (n=dennis@200.32.236.20)
08:08.40*** join/#asterisk Keltus (n=Keltus@about/cooking/nakedchef/beefstew/Keltus)
08:14.42*** join/#asterisk |dennis| (n=dennis@200.32.236.20)
08:53.12*** join/#asterisk Paul_UK (n=foo@email.seatwave.com)
08:53.48Paul_UKhi there, has anyone setup load balancing (not failover) with 2 asterisk servers?  i.e using openser for example?
08:55.20*** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com)
09:06.10Paul_UKhmm, maybe no-one lol
09:08.10tzafrirAnyone probably hasn't set that up.
09:08.24tzafrirHaven't asked No-one yet.
09:09.19JToh, you're finally around, tzafrir
09:10.35JTtzafrir: well, are you around?
09:14.12*** join/#asterisk |dennis| (n=dennis@200.32.236.20)
09:17.17tzafriryes
09:17.57tzafrirJT, yes
09:18.07JTtzafrir: the A500 BRI drivers will use chan_woomera, i've been told
09:18.45coppicewhy are sangoma determined to do the dumbest things? :-\
09:19.03JTyou don't like it?
09:19.50coppicethey say they need to use a proprietary ISDN stack, because nobody wants to fix up the available ones. they have actually gone out of their way to make that hard
09:20.33JThow far removed is the pri stack from bri?
09:20.42coppiceits almost the same
09:20.51Paul_UKok well thanks tzafrir
09:20.58coppicehence, bristuff is minor patches to libpri
09:21.07tzafrirminor?
09:21.30JTyeah, i thought it was weird considering there are plenty of pri stack
09:21.35JTgoing to woomera
09:21.43coppiceyeah, pretty minor in the great scheme of things. most of it provides  the card drivers
09:22.56coppicethere are plenty of crappy PRI stacks. I don't know of one that could get through a compliance suite as it stands
09:23.07coppicea free one, that is
09:23.24JTheh
09:24.45JTgoing to write a good one? ;)
09:25.24coppicebefore the end of this year I intend to have one, unless something gets in the way
09:26.03JTfeature wise will there be any improvements over what we already have?
09:26.24coppiceit will pass compliance suites, for one
09:27.10JTthat's one
09:27.14coppiceIn fact, it will actually come with one
09:29.24coppicethat's the key one. until you get there people will never have stability. all the moaning about BRI would go away with a compliant stack
09:30.07JTstack compliance doesn't necessarily guarantee code stability though
09:30.32coppicestack non-compliance is almost a guarantee of trouble
09:30.53JTi don't doubt it
09:31.02tzafrirboth chan_zap/bristuff and visdn claim to be "compliant", BTW. Not that I have bothered to verify it or check even to what standard
09:32.07coppicethey claim to have gone through some approvals. different thing
09:59.43*** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com)
10:17.53Paul_UKHey guys, i have 3 lines into my office. 1x 2MB:2MB, 1x 1MB:1MB, 1x 8MB:1MB.  They are both loadbalanced at the router and at the switch employ QoS to give VOIP the highest priority.  With Asterisk, since there is only 1 IP (gateway).  Can it effectively use the 4MB of upstream for VOIP?  Also, I have 2 trunks, 1 Call Centre, 1 Main Office.  That if the primary link goes down, the Main Office available lines goes to 0.  Thus making sure that the call ce
10:32.28*** join/#asterisk DragoraN (n=dragoran@217.67.19.74)
10:32.29DragoraNhi all!
10:56.36JTyawn
10:57.13*** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net)
10:57.15kippihey
10:57.31kippiis there away to logon agents without being at there phone
10:57.48*** join/#asterisk stimpie (n=michiel@ip565faf27.direct-adsl.nl)
11:02.51*** join/#asterisk implicit (n=bayan@vc240100.vpn.uci.edu)
11:06.52implicitdenon
11:06.53implicitwats up
11:29.40DragoraNlol
11:30.41JThilarious
11:35.32*** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il)
11:36.31hi365im trying to register my sip url in e164.org and im getting: "Wasn't able to test your route, your system returned the following information: -1|Error 504 - No response was returned from the remote end, this might indicate a problem with a firewall connection"
11:36.40hi365anyone familiar with this error?
11:37.28*** join/#asterisk fusie (n=bla@ip565e1b33.direct-adsl.nl)
11:43.18Paul_UKso no-one knew an answer to my question.  or am i gonna have to set it all up and see lol?
11:43.18JTthe question was too big
11:43.19JTit also got chopped off
11:43.38JTalso, i think you mean Mbit/s, not MB
11:43.39Paul_UKoh thanks
11:43.48Paul_UKi have 3 lines into my office. 1x 2MB:2MB, 1x 1MB:1MB, 1x 8MB:1MB.  They are both loadbalanced at the router and at the switch employ QoS to give VOIP the highest priority.  With Asterisk, since there is only 1 IP (gateway).
11:43.51Paul_UKCan it effectively use the 4MB of upstream for VOIP?  Also, I have 2 trunks, 1 Call Centre, 1 Main Office.  That if the primary link goes down, the Main Office available lines goes to 0.  Thus making sure that the call centre has the priority when taking calls?  Thanks
11:43.57Paul_UKhope thats better
11:45.06JTpretty sure they're not MB
11:45.07JTbut umm
11:45.12JTload balancing might work
11:48.17*** join/#asterisk knarfly (n=knarfly@c-75-74-233-229.hsd1.fl.comcast.net)
11:52.15Paul_UKJT, they're not MB?
11:52.24Paul_UKoh Mbit?
11:52.26Paul_UKthat better?
11:53.05JTyes
11:53.24JTi'm still not sure exactly what you're trying to do
11:53.30JTyou have a few Internet links
11:53.40JTyou have a main office and a call centre
11:53.52JTand 2 trunks, i don't know what type you're refering to there
11:54.52Paul_UKJT, ok alot is I dont know the terminology yet.
11:55.11Paul_UKso bear with me
11:55.46Paul_UKOk, so with 2 main phone numbers, im assuming an inbound and outbound trunk would be associated with each?
11:56.12JTi have no idea how these phone numbers are delivered to you
11:56.21Paul_UKvoip
11:56.25Paul_UKthey arent pstn
11:56.44JTthis company has no pstn?
11:56.48JTnumbers are DIDs
11:56.53Paul_UKyeah
11:57.50JTno normal phone lines of some sort is a bad idea
11:58.14Paul_UKwhere i am currently, its total voip
11:58.33Paul_UKwhere i am moving too, the thought is to keep things the same
11:59.28JTreally bad idea if that's VoIP over Internet
12:01.15Paul_UKwell its an MPLS circuit to the providers network
12:02.00jerPaul_UK, just some background... we did the same thing at our small office. we have redundant links to help assist incase shit happened. as it turned out, they weren't much help, we ended up pulling in a voice t1 line and using the voip trunks strictly for outgoing connections
12:02.08*** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il)
12:03.21Paul_UKjer: I have just over a T1.  2MBit up and down, 1:1 contention
12:03.46JTPaul_UK: i don't think you're understanding, jer said "voice t1"
12:03.54JTnot t1 Internet connection
12:04.03Paul_UKok whats the diff?  is it a leased line?
12:04.12JTit's a voice circuit
12:04.21Paul_UKok well i've never heard of that
12:04.34JTmost businesses with over half a dozen lines use a digital circuit
12:04.40JTusually PRI
12:04.46JTPRI can be T1 or E1
12:04.47jertelephony terminology is fun to learn (and i say that as sarcastically as i can)
12:04.50JTin the UK it would be E1
12:05.06Paul_UKwell the good thing about this, is that im defferring to an outside company to consult me.  If it ends up being crap, then I can blame them.
12:05.19JTsome countries, including the UK have BRI available too
12:05.24JTbasic rate interface
12:05.24jerPaul_UK, if it ends up being crap, you have a loss of revenue no doubt
12:05.27jerwhich is not a good thing
12:05.29JTas opposed to primary rate
12:05.49JTbasic rate is 2 voice channels, and a signalling channel
12:05.59Paul_UKjer, yeah i agree on that.  but we have a backup in last, should things go wrong.  still the phone system wont be up and running until 30th, so i have time to sort out any issues.
12:06.02JTprimary rate t1 is 23 voice channels and a signalling channel
12:06.12JTprimary rate e1 is 30 voice channels and a signalling channel
12:07.14Paul_UKok well i will talk with my provider on tuesday and see what they say
12:07.52tzafrirIf you're from the UK, you have E1...
12:08.24JTmpls directly to provider is better than VoIPoI
12:08.57Paul_UKoh i was gonna say mpls
12:09.10Paul_UKits kinda like frame-relay?  i think thats what we have
12:09.16JTno...
12:09.37Paul_UKok well im wrong again lol..  but mpls is decent enough?
12:09.47JTthat really depends who runs it
12:10.05Paul_UKwow so many variables, well this is my first implementation.  if it fucks up, then it does.
12:10.06JTat least telcos with their PRIs are usually fairly likely to have reliability downpat :)
12:11.14*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
12:11.41Paul_UKok well with that out of the way, what about loadbalancing, will it fly?
12:12.10JTdunno
12:12.17JTare they bonded or just load balanced?
12:12.23*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
12:12.30Paul_UKi would say load balanced
12:12.58JTif the load balancer operates correctly with respect to sip and rtp, then it could work
12:14.24*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
12:16.06*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
12:21.14*** join/#asterisk saftsack (n=saftsack@pD9E06297.dip.t-dialin.net)
12:21.29*** join/#asterisk friedrich| (n=friedric@e177245074.adsl.alicedsl.de)
12:25.01*** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr)
12:30.32*** join/#asterisk zotz (n=zotz@24.244.163.157)
12:31.14*** part/#asterisk zotz (n=zotz@24.244.163.157)
12:35.35*** join/#asterisk jhiver_ (i=jhiver@165-242.206-83.static-ip.oleane.fr)
12:36.32jhiver_hi all
12:36.59jhiver_can anybody let me know if you see anything wrong in my sip.conf? When i try to make a test call i get a 404 not found
12:37.02jhiver_http://pastebin.ca/629552
12:37.57jhiver_i've tried to set core set debug and core set verbose but i don't see anything on the CLI
12:42.09*** join/#asterisk DrukenLPY (n=jdumais@CPE000e08cb2a29-CM00137189cb0c.cpe.net.cable.rogers.com)
12:42.14DrukenLPYmorning everyone
12:42.18jhiver_hi
12:46.53jhiveris somebody awake? i need a pair of eyeballs to look at my dialplan because asterisk returns a 404 when i don't think it should, so i'm prolly doing something wrong
12:47.11DrukenLPYpastebin it
12:48.08jhiverok
12:48.52jhiverhttp://pastebin.ca/629561
12:50.18JTwhich pastebins do you want us to look at?
12:50.34jhiverwell the first is my sip.conf
12:50.39jhiverthe second my extensions.conf
12:51.19jhiveri think the sip.conf is ok
12:51.56JTsorry, what are you using to call, what are you trying to call?
12:52.00jhiversip debug says "Looking for 0017400062932847 in routes (domain 91.121.9.144)", and if leave just the following line:
12:52.04jhiverexten => 0017400XXXXXXXXX,1,Macro(balkana,${EXTEN:7})
12:52.14jhiverthen it doesn't match and i get a 404
12:52.18jhiverok
12:52.31JTyour match is incorrect
12:52.36JTadd a _ at the start
12:52.36jhiverI'm using a piece of software call voipswitch to place the call to the asterisk box
12:52.44JTi see
12:52.49jhiver0017400062932847
12:52.53jhiver0017400XXXXXXXXX
12:52.58JTno.
12:53.00jhiverseems to match to me
12:53.02JT_0017400XXXXXXXXX
12:53.06jhiveraaaaah =)
12:53.07JTwell you're mistaken
12:53.09jhiversilly me =)
12:53.13jhiverthx =)
12:53.17JTno underscore, no pattern match
12:53.26jhiveryes yes of course
12:53.37jhiveri'm all rusty with asterisk dialplans! thanks a bunch
12:53.48JTno probs
12:54.28jhiveris 'DBput' deprecated in 1.4 ?
12:54.54jhiveri have this now : No application 'DBput' for extension (macro-balkana, s, 3)
12:56.58jhiveraaah looks like they have become functions now?
12:57.06jhiverjeez, i'm real rusty =)
12:59.32DrukenLPYi'm having issues with realtime and odbc today... really annoying
13:12.41*** join/#asterisk _DAW (n=_DAW@adsl-074-185-215-060.sip.msy.bellsouth.net)
13:16.06*** join/#asterisk Strom_M (n=strom@s142-179-221-179.ab.hsia.telus.net)
13:16.43*** join/#asterisk huey23 (n=huey23@64-184-9-88.bb.hrtc.net)
13:22.42DragoraNjhiver
13:22.43DragoraNRelease 1.2 and later
13:22.43DragoraNDBPut(family/key=${foo})    (deprecated)
13:22.43DragoraNSet(DB(family/key)=${foo})  (new syntax)
13:22.43DragoraNRelease 1.4 and later
13:22.43DragoraNSet(DB(family/key)=${foo})
13:22.51DragoraNsorry for flood
13:26.22jhiverjeeez
13:26.27jhiverRandom() is deprecated too?
13:26.36jhiverdamn, the new syntax is ugly as hell =)
13:26.51DragoraNjhiver http://www.voip-info.org
13:28.00DrukenLPYjhiver: i agree with ya... it's very ugry
13:28.21JTit's more logical to use Set for everything than to have pointless little apps
13:28.32JTto set a variable...
13:31.05DragoraNi agree with JT  :)
13:31.28DragoraNBFU's will never understand coding style..
13:31.30DrukenLPYi agree as well, i just don't like the new syntax...
13:31.42DrukenLPYbfu's?
13:32.25DragoraNbfu - normal user :)
13:32.31DragoraNconsumer ;)
13:32.44DrukenLPYhow in hell does BFU stand for normal user or consumer?
13:33.07DragoraNbloody f... user
13:33.22DrukenLPYahh, ok
13:33.37DragoraNan abbreviation for a geographically limited term for an unskilled computer user, the Bloody Fucking User or Brain Free User or Basic Function User. This term appears to be used by non-native English speakers only in the Czech Republic, Slovakia and neighbouring countries.
13:33.57DragoraNcz/sk only.. :)
13:34.14DrukenLPYwell, that would make sence... since i'm canadian.. :)
13:34.23DrukenLPYEH!
13:34.23jhiverlads, is there an equivalent to dbdeltree on the CLI?
13:36.19DragoraNjhiver: dont be lazy :)
13:36.25DragoraNjhiver: database deltree
13:36.34jhiverlet me see
13:36.38DragoraNEtch*CLI> database deltree
13:36.38DragoraNUsage: database deltree <family> [keytree]
13:36.38DragoraN<PROTECTED>
13:36.38DragoraNin the Asterisk database.
13:37.17jhivercool
13:37.48jhiverso i can use it from the shell with the '-rx' flags... nice
13:38.03DragoraNawesome! :)
13:38.43jhiversorry about this, i haven't touched asterisk since 1.0.9, it's changed quite a bit =)
13:39.20DragoraNjhiver: need a bridge course?
13:39.40jhiverheh, i think you've just made my day =)
13:40.13DrukenLPY1.0.9 to 1.4.... you might as well forget everything you learned, and start over fresh :)
13:41.51DragoraNonce upon a time, there was one voip
13:43.04DragoraNdoes anyone use iaxtermination ?
13:44.35*** join/#asterisk Strom_M (n=strom@s142-179-221-179.ab.hsia.telus.net)
13:44.44*** join/#asterisk d00gster (n=doughant@bas1-toronto12-1128667111.dsl.bell.ca)
13:50.25*** join/#asterisk ManxPower (n=manxpowe@31.sub-70-223-183.myvzw.com)
13:57.02DrukenLPYwell, looks like asterisk has been playing nice for about 2 hours
13:59.43*** part/#asterisk Strom_M (n=strom@s142-179-221-179.ab.hsia.telus.net)
13:59.57*** join/#asterisk Strom_M (n=strom@s142-179-221-179.ab.hsia.telus.net)
14:00.14*** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com)
14:00.50*** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2)
14:02.12*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
14:05.16*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
14:07.55*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
14:09.06*** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com)
14:11.00*** join/#asterisk saftsack (n=saftsack@pD9E06297.dip.t-dialin.net)
14:12.33*** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-241-176-115.buff.east.verizon.net)
14:23.22*** join/#asterisk Strom_C (n=strom@s142-179-221-179.ab.hsia.telus.net)
14:34.26*** join/#asterisk mightnare (n=mike@p3203-ipad03motosinmat.mie.ocn.ne.jp)
14:34.47*** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com)
14:35.47*** join/#asterisk Ebola (n=Ebola@host86-138-208-67.range86-138.btcentralplus.com)
14:36.45*** join/#asterisk cayorde (n=flexable@host245-111-dynamic.17-87-r.retail.telecomitalia.it)
14:37.43*** join/#asterisk Jiboneus (n=Jibone@124.82.84.94)
14:39.19jhiveryo
14:39.22jhiveragain =)
14:40.35jhiveris there a known way to force asterisk to stay out of the media path when connecting two SIP legs? I don't have any natted devices, is canreinvite=yes sufficient? I've put this option under [general] but i'm still seing some "g729 frames" errors when really, asterisk should never set itself as an endpoint...
14:42.12Strom_Cjhiver: asterisk is not a sip proxy
14:42.42Strom_Casterisk will put itself in the media path first and then only drop out if the other endpoints can talk directly
14:42.52Strom_Chence "back to back user agent"
14:44.10kippiis there away to logon agents without being at there phone
14:45.14*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
14:45.29Nuggetof course.  just use addqueuemember
15:05.46*** join/#asterisk mvanbaak (n=mafkees@vanbaak.xs4all.nl)
15:07.09*** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com)
15:11.04*** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
15:11.04*** mode/#asterisk [+o drumkilla] by ChanServ
15:17.01*** join/#asterisk fnordus (n=dnall@24.84.160.227)
15:29.36*** join/#asterisk Greek-Boy (n=Greek-Bo@196.45.144.42)
15:35.39*** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il)
15:37.53hi365can i set a range in the dial plan? i currently have this : exten => s,n,Gosubif($["${CHANNEL:4:1}"="9"]?set1800|1)
15:38.15hi365how can i make it "grater than 9 but less then 13"?
15:38.30hi365of =9-13
15:38.36hi365of=or
15:39.01*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
15:40.54*** join/#asterisk fakhir (n=Fakhir@unaffiliated/fakhir)
15:49.50*** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com)
15:51.24drumkillahm ... $[ $["${CHANNEL:4:1}" > "9"]$["${CHANNEL:4:1}" < "13"] = 11]
15:51.50drumkillaAEL would make that a whole lot easier, heh
15:52.07drumkillaand it would probably be most readable in conf format if you just break it up over a few lines
15:58.23*** join/#asterisk op3r (n=Op3r@121.97.193.145)
16:00.24*** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust826.leed.cable.ntl.com)
16:01.34hi365drumkilla: true but im planing on doing agent logins and its like 60 agents
16:02.38hi365drumkilla: whats the "=11"?
16:03.52drumkillawell, it's two expressions, and then it's checking to see if both evaluated to 1
16:03.58drumkillaand if so, the whole thing evaluates to 1
16:04.00drumkilla(i think)
16:09.35hi365drumkilla: ah! thanks ill try it
16:26.44*** join/#asterisk Delvar (n=Delvar@host-83-146-53-46.bulldogdsl.com)
16:27.42*** join/#asterisk saftsack (n=saftsack@pD9E06297.dip.t-dialin.net)
16:40.09*** join/#asterisk Juggie (n=Juggie@CPE00c049d9f271-CM000a73a18a20.cpe.net.cable.rogers.com)
16:51.18*** join/#asterisk |dennis| (n=dennis@200.32.236.20)
16:55.59Hmmhesaysoh I like vmware
17:00.16*** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net)
17:13.03*** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com)
17:18.33*** join/#asterisk saftsack (n=saftsack@pD9E06297.dip.t-dialin.net)
17:25.13*** join/#asterisk ukris (n=ukris@p1016-ipad605marunouchi.tokyo.ocn.ne.jp)
17:26.36ukriswhat is the default DTMF mode for Asterisk Voice Mail System .I am using asterisk 1.2.13
17:27.08Strom_CDTMF is determined by the channel driver, not by the voicemail application
17:28.26ukristhanks Storm_C .I tried to channge the DTMF settiings on the sip.conf from RFC to INBAND .Dosent seem to work with the GXP2000 .
17:28.35*** part/#asterisk hi365 (n=hi365@mail.pcgeula.co.il)
17:33.23*** join/#asterisk ukris (n=ukris@p1016-ipad605marunouchi.tokyo.ocn.ne.jp)
17:37.23*** join/#asterisk icanicant (n=icanican@dsl-217-155-248-73.zen.co.uk)
17:39.10*** join/#asterisk zotz (n=zotz@24.244.163.157)
17:43.56*** join/#asterisk [hC] (n=hardcore@pool-71-112-142-43.sttlwa.dsl-w.verizon.net)
18:05.59*** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il)
18:10.45hi365any enum guru's in the house?
18:11.03hi365im getting error 504 when i try to register my sip url
18:26.07*** join/#asterisk adderd (n=adderd@ppp-70-226-87-198.dsl.klmzmi.ameritech.net)
18:26.36adderdhello, I've got a quick question (hopefully)
18:26.46adderdDo either IAXTEL or DUNDi exist any longer?
18:26.53adderdboth websites are missing in action
18:33.52*** join/#asterisk zeedo (n=zeedo@reboot-robot.net)
18:46.59*** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca)
18:47.02adderdis no one alive in here?
18:48.35hi365not today
18:53.29*** join/#asterisk hwtrap (n=asdasd@200-101-63-246.cpece705.dsl.brasiltelecom.net.br)
18:55.39*** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org)
18:57.18adderddoes anyone know the where abouts of dundi or iaxtel?
18:58.48*** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga)
19:18.47icanicanti see that inbound audio from my SIP provider is coming in to a range of UDP ports around 5000 to 5100. it's being delivered directly to the local SIP phones from the ITSP rather than from Asterisk (i.e. from a range of external IP addresses). Does this mean I have to open a large range of UDP ports or can I get Asterisk to sit in the middle and deliver the packets from the Asterisk box?
19:22.16*** join/#asterisk Gouroutrash (n=x@ACaen-151-1-43-85.w86-215.abo.wanadoo.fr)
19:22.35Gouroutrashhello
19:23.01x86icanicant: you can make Asterisk talk IAX2 to your ITSP, and only use a single port for signalling and RTP
19:23.12x86if your ITSP doesn't suck
19:23.36x86Gouroutrash: heya
19:24.12Gouroutrashanybody have a solution for kill all the beginners who use trixbox in the place of Asterisk ?
19:24.14Gouroutrash:)
19:25.12icanicantx86: thanks. failing that, do the local SIP extensions need to be open for ports 1000 to 65535 or a smaller range? google tells me all ports but in practice it looks like i'm getting them very close to the SIP 5060 port num.
19:25.15Gouroutrashi'm bored when i look questions on forums
19:26.56*** join/#asterisk xuser (i=xuser@unaffiliated/xuser)
19:27.28xuserHi, is there a open source billing software for VoIP Gateways?
19:28.00*** join/#asterisk xrg_ (n=panos@88.218.82.82)
19:28.35xrg_Juggie: hi..
19:28.46Gouroutrashyes, "starshop" is opensource
19:29.16Gouroutrashi just know the name, never test it
19:29.52xrg_Gouroutrash: I have: starshop is a minimal callshop implementation.
19:30.11Gouroutrashok :)
19:31.42xrg_In a few months, hopefully, you will see the callshop modules for asterisk2billing.. They will be much more feature-rich..
19:36.24*** join/#asterisk |dennis| (n=dennis@200.32.236.20)
19:39.33*** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com)
19:41.25DragoraNwhat is better, asterisk or callweaver?
19:41.37DragoraNi like sqlite, but * is *
19:50.18x86mysql > sqlite
19:50.39x86callweaver has most of the features of asterisk, and more
19:50.48x86like IAX2 / SIP jitterbuffers
19:51.00x86and T.38 support
19:54.36DragoraNso callweaver?
19:54.42DrukenLPYanyone had a problem with database calls getting all screwed up after a while?
19:55.54x86DrukenLPY: you mean CDR records stored in a database?
19:56.00x86DrukenLPY: depends on what your needs are
19:56.04x86err
19:56.10x86DragoraN: depends on what your needs are
19:56.31x86if you need generic jitterbuffer support in SIP and IAX2, and are doing T.38 faxing, use callweaver
19:56.45x86if you want supportability and stability, get asterisk
19:56.55DragoraNx86: i want to use smart, extensible, opensource, well-documented, fast PBX with SIP, IAX, ...
19:57.28x86well documented probably equates to asterisk
19:57.59DrukenLPYx86: nope... i mean odbc, and database gets... realtime, dialplan gets, everything
20:02.04*** join/#asterisk op3r (n=Op3r@121.97.193.145)
20:05.09x86DrukenLPY: never used ODBC for realtime, and wouldn't recommend doing so
20:10.26jj56x86: why not? not using realtime, or just don't use it with ODBC?
20:11.30jj56x86: i'm planning on using realtime with mysql, is that stable?
20:11.35x86sure is
20:12.27jj56how realtime is realtime.. do the updates to the dialplan happen during calls? or periodically?
20:12.54x86all the time
20:14.50jj56sounds good, do you know of any limitations such as maximum extensions / dialplan?
20:15.27jj56i'm planning on having multiple customers design their own IVR using a webbased interface..
20:15.37jj56so the server would host 'multiple dialplans'
20:16.08jj56that could result in a massive dialplan
20:16.42jj56are there any limits to that?
20:18.02DrukenLPYx86: why no odbc for realtime? it's worked perfect for months.... and all of a sudden went screwy
20:21.19adderdanyone alive now?
20:23.30jj56lemme check
20:23.37jj56yeh :)
20:25.03x86DrukenLPY: that's why i wouldnt recommend it
20:25.18DrukenLPYwhich is?
20:25.37adderdso whats the deal with DUNDi and IAXtel
20:25.37x86ODBC
20:25.40adderdare they both dead?
20:25.52x86adderd: IAXtel is run by someone at digium now
20:26.00x86you have to email for an account, last i checked
20:26.11x86mark gave me the email addy once but i lost it
20:26.18adderdwell the website is dead anyway
20:26.23x86yeah
20:26.24adderdsame thing with Dundi
20:26.48x86dundi is a dialplan function
20:26.54x86"application"
20:27.01adderdthe website for dundi is non existant now
20:27.05adderdthat's what I'm saying
20:27.11adderdthe directory is totally empty
20:27.27adderdmakes it more difficult to get information on it
20:28.41*** join/#asterisk Greek-Boy (n=Greek-Bo@196.45.144.42)
20:37.12*** join/#asterisk santiago (i=santiago@debian/developer/santiago)
20:47.06*** join/#asterisk kirberich (n=robert@i538719C1.versanet.de)
20:47.09kirberichhi
20:48.32cy-sup kirberich
20:48.42kirberichi have a hfc pci card here, and i'm trying to use it in nt-mode with an isdn-telephone attached. softwarewise everything seems finde, and the cabling should be correct too, but i can't get any interaction between computer and phone
20:49.12jj56still no solution eh? sorry to hear that..
20:49.36kirberichjj56, yeah, i was about to throw the damn thing out of the windows yesterday ;)
20:49.52jj56why don't you?
20:50.09kirberichbut well, i know that it has to work somehow, i just need to find a person who's done it before ;)
20:50.41kirberich(getting it to work, not throwing it out the window)
20:51.12jj56sorry, i threw a lot of stuff out of the window, but no hfc :)
20:55.36jj56in fact, lets throw this pc out of the window
21:00.21*** join/#asterisk data23 (i=data@92.b6.3845.static.theplanet.com)
21:00.22*** join/#asterisk |dennis| (n=dennis@200.32.236.20)
21:11.01*** join/#asterisk gfgfgfgf (n=davidh@vc-196-207-45-253.3g.vodacom.co.za)
21:11.21gfgfgfgfwhere does one start?
21:12.20gfgfgfgfcan antone please help  me i need to know what i need to get asterisk working?
21:13.13DragoraNgfgfgfgf: please tell us your problem first
21:13.52DragoraNgfgfgfgf: what do you need? x86 computer
21:15.18gfgfgfgfi am new tried installing asterisk without any harware to no avail,\ need to know where i can find a list of software asterisk needs to work ,should this not be deferent for every tipe of linuc beeing fedora,suse or debian!?
21:15.24DragoraNgfgfgfgf: http://www.voip-info.org/wiki/view/Asterisk+introduction
21:16.07DragoraNgfgfgfgf: try ubuntu
21:17.16hypa7iagfgfgfgf: it's fairly different, you'll want a how-to for your distro
21:18.06*** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.16)
21:18.27DragoraNgfgfgfgf: http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Ubuntu here you have cookbook for asterisk
21:18.33DragoraNgfgfgfgf: on ubuntu
21:18.33gfgfgfgfno what dependesies asterisk requires?
21:18.44hypa7iagfgfgfgf: have a look at those links
21:18.56DragoraNtry ubuntu.. dpkg will handle dependencies for you
21:19.01hypa7iagfgfgfgf: what distro have you tried it on?
21:19.11gfgfgfgfsuse 10.2
21:19.17hypa7iaeep
21:19.18hypa7iayeah
21:19.31DragoraNgfgfgfgf: ok.. you install asterisk using yast
21:19.36hypa7iatry ubuntu or CentOS, most people are on those
21:19.45gfgfgfgfno wget
21:20.10hypa7iagfgfgfgf: do you have a reason you need to custom-compile it rather than using a package?
21:20.38gfgfgfgfyes want to use asterisk v1.4
21:20.54DragoraNthere are packages for 1.4
21:21.11gfgfgfgfmeaning?
21:21.50hypa7iameaning you don't need to compile it
21:22.06hypa7iayou can use a package that someone else has compiled for you :)
21:22.25hypa7iakind of like an msi on windows
21:22.51*** join/#asterisk |dennis| (n=dennis@200.32.236.20)
21:22.55DragoraNgfgfgfgf: please, read briefly the at least first link i gave you...
21:23.58gfgfgfgfi read the handbook but cant test the dial plan unless asterisk starts
21:25.00gfgfgfgfwhere can i find a list of  dependecies needed by asterisk
21:27.27*** join/#asterisk Mahmoud (n=Mahmoud@unaffiliated/mahmoud)
21:27.29Mahmoudhello
21:27.39Mahmoudi'm using sipura 3000. how to skip * in my dialplan?
21:27.50Mahmoudnot really related to asterisk, but i'm sure many people here use it
21:28.07Mahmoudi want to skip * and #
21:33.47*** join/#asterisk justdave (n=dave@unaffiliated/justdave)
21:33.57Strom_CMahmoud: what do you mean "skip * and #"?
21:34.16MahmoudStrom_C: so i can dial a number that includes #
21:34.24Mahmoudsipura 3000 has a special meaning of #
21:34.41Strom_CMahmoud: as does all of the rest of telephony
21:34.46Strom_Cdon't use # as part of a telephone number
21:34.56Mahmouddamn. it's already used by my ISP
21:35.03Strom_C# has meant "I am finished dialing; place the call now" since 1965
21:35.07Mahmoudfor special services, such as locking the phone down
21:35.13Mahmoudi know what # means
21:35.23Mahmoudi already skipped #, but forgot now
21:35.47Mahmoudproblem is that, i can't dial special numbers that include #.. just because spa has special understanding
21:36.13Strom_Cgive me an example
21:36.40hypa7iagfgfgfgf: if you install from a package it will load all the dependencies you need
21:36.44hypa7iaand then it will start
21:36.50hypa7iagive that a try :)
21:37.20MahmoudStrom_C: for example, to lock my phone, i dial #33*1234#
21:37.27MahmoudStrom_C: where 1234 is my pass key
21:37.55*** join/#asterisk legis (i=legis@unaffiliated/legis)
21:37.58hypa7iagfgfgfgf: do you know how to use YAST?
21:38.34Strom_CMahmoud: that
21:38.42Strom_Cthat's kind of stupid
21:38.49Strom_Cdumb ISP for the lose
21:38.49Mahmoudwell, this is my ISP
21:39.11Mahmoudaghhh
21:39.24Mahmoudi already did it and it was working.. but today i removed it by mistake and saved the config.. damn it
21:39.29legisHi, can * be use as a softswitch and billing VoIP platform?
21:39.46MahmoudStrom_C: if you know a way to skip these characters in spa 3000 tell me..
21:40.06*** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal)
21:40.17hypa7ialegis: as in for an ITSP?
21:40.30*** join/#asterisk MrMister2 (n=mrmister@89-180-119-4.net.novis.pt)
21:40.57legishypa7ia: yeah
21:41.03legissomething like mera
21:41.42hypa7iai don't know what mera is, but there are definitely people using asterisk as an ITSP platform
21:41.47hypa7iabut it'll take some config work
21:42.47legisbasically what i want is to centralized various voip gateways through a softswitch.
21:43.02legiswondering if * is the a good tool for the job.
21:43.31hypa7iacentralize as in manage the billing for the other softswiches or as in replace them? :)
21:43.59legisreplace
21:44.46hypa7iai think it would do what you want. but without more detail i can't say for sure
21:45.30*** join/#asterisk jpe-nyc (n=jpe-nyc@p77-37.acedsl.com)
21:45.43xuserok
22:00.29Juggiehypa7ia!
22:00.43hypa7iaJuggie!
22:00.53hypa7iadude are you still in ottawa?
22:01.02Juggieyeah
22:01.22hypa7iacool
22:01.25hypa7iai'm there now
22:01.33Juggiei live in barrhaven now, bought a house
22:01.35*** join/#asterisk easimon (n=easimon@baghira.kawo2.RWTH-Aachen.DE)
22:01.42hypa7iacoolio
22:02.03Juggiewhat are you doing back in ottawa
22:03.10hypa7iasome work for my dad
22:03.41hypa7iayou still at irs-north?
22:03.42hypa7ia:)
22:05.09Juggieyep
22:05.17Juggiestill working for the feds in the market
22:07.30Mahmoudwhat are the cases when I get circuit busy ?
22:07.45Juggiewell, when its busy.
22:08.40Mahmoudwhen the line is used by others?
22:08.51Juggiethat would be one case yes
22:08.58Mahmoudanother case is?
22:09.41Juggiewell, it could be a group of lines thats entirely busy
22:10.24Juggieif your on a t1, its possible your telco could return a circut busy/congestion to you
22:10.30Juggieif they are unable to deliver your call
22:10.31Juggieetc
22:13.00Mahmoudusing a single pots line
22:13.04Mahmoudand it tells me busy
22:13.08Mahmoud100% sure no one uses the line
22:13.23Mahmoudwill restart my SPA for final check..
22:13.44_DAWMahmoud: are you terminating the SPA on an * box?
22:14.12Mahmoudspa is connected to:
22:14.13Mahmoudanalog phone
22:14.13Mahmoudasterisk
22:14.13Mahmoudpots phoneline
22:20.27Mahmouddamn
22:20.33Mahmoudno one using any line, and yet, congested
22:22.42_DAWare you getting congestion from the sipura directly?
22:26.42*** join/#asterisk zotz (n=zotz@24.244.163.157)
22:26.47Mahmoud_DAW:  -- SIP/pots-086f1000 is circuit-busy <---- SIP/pots is my sipura account
22:28.24JTsure you can probably try asterisk out via packages, some are better than others
22:28.30JTbut generally you'll want to compile it
22:29.38Mahmoudit was working fine, i screwed things up :(
22:36.59*** join/#asterisk CrazyTux[m] (n=CrazyTux@cpe-76-175-64-30.socal.res.rr.com)
22:39.11russellbi'm just curious, what makes asterisk tough to deal with as a packaged app?
22:39.29russellbit's a common opinion, just hadn't spent much time thinking about why
22:39.44InnatechUsually the provided dialplans are somewhat labrynthine and hard to adjust if you want to do anything unanticipated.
22:40.21russellbwell, i mean from an rpm, deb, or your package manager of choice
22:40.27JTrussellb: they're always so old
22:40.32russellbnot necessarily "packaged" as in with a gui ..
22:40.34Innatechah.
22:40.39russellbJT: ah ..
22:40.43JTand sometimes you may wish to add patches before you compile
22:41.07JTand sometimes they come with on crack default configs
22:41.14russellbkind of like how many people are with their kernel?
22:41.15JTso it's a number of things i guess
22:41.51russellbfair enough, just curious.
22:42.14DrukenLPYrussellb: it's a control issue :)
22:42.33DrukenLPYlike how most drivers are horrible passengers....
22:42.52russellbha, nice
22:43.31*** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net)
22:44.00russellbManxPower: the Digium PBX is on 1.4 now :-p
22:49.47CoolGuy21digium pbx?
22:50.00CoolGuy21asterisk
22:50.12JTCoolGuy21: the digium pbx
22:51.14DrukenLPYdigiums phone system
22:52.31russellbyeah, the main asterisk box at digium ... heh
22:59.25Mahmoudhmm
23:06.25*** join/#asterisk mtaht4 (n=m@cpe-74-76-23-86.nycap.res.rr.com)
23:09.45MahmoudStrom_C: spa says that the pots line is idle.. how come it says its busy? weird :/
23:10.35Mahmoudrunning asterisk -rvvvvvvvvvvvvvvvvvvv
23:10.50Mahmoudwhen I call any phone in the pots network, i get this error: http://pastebin.ca/630156
23:11.46MahmoudSIP/pots is SPA's account regarding its POTS interface
23:12.17Mahmoudthe interface connected to the ISP
23:13.39JTpretty useless cli output
23:13.44JTat least turn on sip debug
23:13.51JTand does the spa have its own log?
23:14.08Mahmoudnope
23:14.15Mahmoudcouldn't find spa logs
23:19.37*** join/#asterisk hohum (n=dcorbe@dhcp64-134-231-245.shs.nyc.wayport.net)
23:21.24MahmoudJT, does this say anything usefull? http://pastebin.ca/630164
23:21.44Mahmoudjt, asterisk is 10.1.0.1 and spa device is 10.1.0.3
23:21.59JTcope
23:22.01JTnopw
23:22.25JTwhy would you paste just one message in the sip dialogue? what a waste of time
23:22.30JTfull call or don't bother
23:23.05Mahmoudany easy way to select the whole text?
23:23.13Mahmoudscrolling down and copy pasting is a bit tough
23:23.49JTyou could extract it from the full log if it's any easier
23:24.09Mahmoudwhere is it located?
23:24.34JTif it's enabled, /var/log/asterisk/full
23:27.01Mahmoudhow to enable it? :/
23:27.07Mahmoudnvm
23:30.57DrukenLPYhttp://www.pastebin.ca/630171
23:32.31MahmoudJT, http://pastebin.ca/630172
23:35.29Mahmoudhmmm
23:39.13MahmoudJT, i think you already answered a similar question in mail-archive.com
23:39.27MahmoudJT, i think this is your reply http://www.mail-archive.com/asterisk-users@lists.digium.com/msg16441.html
23:39.40*** join/#asterisk ManxPower (n=manxpowe@31.sub-70-223-183.myvzw.com)
23:41.05*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
23:41.24puzzledevening all
23:42.14Juggiepuzzled
23:42.36JTMahmoud: that's not me
23:42.47MahmoudJT, i see
23:43.00Mahmoudjt, i disabled qualify lines in sip.conf and still doesn't work..
23:45.24JTMahmoud: you did not paste some important stuff from later in the sip dialogue
23:45.37JTi really do mean paste everything
23:45.46JTnot what you think is important
23:46.04JTeverything from just before you make the call, to just after it fails on the asterisk console
23:46.14Mahmoudok
23:46.18JTotherwise you are wasting time
23:49.24MahmoudJT, full log http://pastebin.ca/630198
23:49.46MahmoudJT, the starts by starting asterisk till the call is end
23:50.52*** join/#asterisk |dennis| (n=dennis@200.32.236.20)
23:53.12JTthe spa probably doesn't know how to route that call
23:53.32JTalso, looks like you're misusing qualify= in your configs, as there's tonnes of errors about them
23:55.00MahmoudJT, hmm weird.. will check if i was editing wrong file or no
23:55.04Mahmouds/no/not/
23:57.11MahmoudJT.. ohh sorry, wrong log file
23:59.19MahmoudJT, currect log file http://pastebin.ca/630211
23:59.29DrukenLPYcan someone help me with a realtime problem ?
23:59.40MahmoudJT the previous log file wasn't updated.. it was a copy of a previous one

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.