00:00.12 | Berny_ | getting the dial plan and all the regional settings 100% was a pest |
00:00.58 | Berny_ | 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 |
00:01.08 | Berny_ | teething problems I wrote it downt o |
00:01.10 | Berny_ | to eve |
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00:02.57 | x86 | perfect MoH material here |
00:03.37 | Innatech_away | "press one for WICKED TUNES." |
00:04.02 | x86 | heh... |
00:04.26 | x86 | customer calls for support, and while sitting in the queue they hear "HOLY SMOKE! HOLY SMOKE! THIS IS NO JOKE!" |
00:04.30 | x86 | roflmao |
00:04.57 | Berny_ | hehehe |
00:06.45 | Berny_ | x86, just your beer comment, best place is belgium |
00:07.10 | x86 | as long as you didn't try to say best place for beer is london ;) |
00:07.11 | Berny_ | antwerp has a beer festival once a year, if you are around in europe, it's worth a look |
00:07.25 | Berny_ | nup |
00:07.25 | x86 | i'm a big fan of german beer, like Erdinger |
00:07.35 | Berny_ | ah ok |
00:07.39 | x86 | Honey Weiss |
00:08.54 | *** join/#asterisk powerkill (n=powerkil@84.205.154.247) |
00:09.01 | Berny_ | I think I had some of that in belgium |
00:11.11 | Berny_ | 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) |
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00:57.11 | *** join/#asterisk RazaMetaL (n=razameta@200.93.220.27) |
00:57.14 | RazaMetaL | hi guys |
00:57.26 | RazaMetaL | does any one using astribank ? |
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02:08.36 | MoutaPT | x86 i'm not using ztdummy , i believe... |
02:09.02 | MoutaPT | i'm wondering if it is supposed to appear zap/pseudo on channels when using Meetme appliaction |
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03:03.51 | killfill | i 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? |
03:03.55 | killfill | how do i reset this? |
03:05.00 | *** join/#asterisk grimbeans (n=audiokry@202.88.74.13) |
03:05.09 | grimbeans | hi fellas |
03:05.29 | grimbeans | I'm new to asterisk and have a question.. could someone please help me? |
03:05.47 | grimbeans | ok here goes.. :p |
03:05.57 | grimbeans | I'm starting up a small internet telephony service locally.. |
03:06.04 | grimbeans | I'm going to be using 16 analog trunk lines initially by way of analog FXO gateways |
03:06.12 | grimbeans | the numbers will be 234-1001 through 234-1016 |
03:06.18 | grimbeans | I want all my customers to call just 234-1001 |
03:06.23 | grimbeans | as each call is received, I want them to be automatically transferred (call distributed?) to one of the other 15 lines |
03:06.30 | grimbeans | so that the original 234-1001 is constantly free and open to new calls |
03:06.42 | grimbeans | does AsteriskNOW's automatic call distribution feature allow me todo this? |
03:06.56 | JT | umm |
03:07.08 | JT | you should NOT be using analogue lines to run an ITSP |
03:07.14 | grimbeans | i know.. T_T |
03:07.28 | grimbeans | but the local telco provides T1 at $8,000 a month!!!! |
03:07.33 | grimbeans | damn monopoly :( |
03:07.50 | Hymie | grimbeans: you're paying for call transferring too? |
03:07.52 | JT | for voice service? |
03:08.03 | Hymie | and, that would be cumbersome |
03:08.26 | grimbeans | the local telco apparently does not provide call forwarding on busy |
03:08.30 | Hymie | why not just pay for a ring down service from the telco |
03:08.39 | grimbeans | and i think they don't distinguish between voice/data T1 here |
03:08.44 | grimbeans | a ring down service? |
03:08.47 | Hymie | yes |
03:08.51 | Hymie | it's a fairly standard thing |
03:08.52 | grimbeans | what is that? (i'm noob) :( |
03:08.54 | Hymie | you dial one number |
03:09.09 | Hymie | and the telco sets it up so it will find any free line and ring that, in the ringdown group |
03:09.23 | Hymie | that's how most companies handle having 5 or 10 lines |
03:09.35 | grimbeans | that would be great :D |
03:09.52 | grimbeans | I just hope my local telco is willing to provide that service.. |
03:09.52 | coppice | its more commonly called a hunting group |
03:09.55 | Hymie | I'm not sure if there is an upper limit on the number of ring down lines allowed |
03:10.17 | Hymie | coppice: probably I'm speaking local telco lingo.. they call it ring down lines here :Þ |
03:10.26 | grimbeans | ic ic.. :) |
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03:10.39 | grimbeans | I guess I ought to call the telco on Monday to get answers.. |
03:10.47 | Hymie | indeed |
03:10.51 | Hymie | you pay for the service though |
03:10.59 | Hymie | not sure what the extra $$$ is |
03:11.06 | Hymie | of course |
03:11.13 | grimbeans | hopefully it's not too crazy a price :p |
03:11.20 | Hymie | are you planning to use voip for the dialout service? |
03:11.25 | Hymie | is this part of the asterisk plan? |
03:11.25 | grimbeans | yep :) |
03:11.32 | Hymie | then why not use voip for incoming? |
03:12.07 | Hymie | it's probably cheaper, althugh you'll need more internet bandwidth |
03:12.25 | grimbeans | hmm.. |
03:12.45 | Hymie | that'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 |
03:13.07 | Hymie | you're going to need fibre or host this at a local isp with a GOOD backbone |
03:13.14 | Hymie | otherwise, your customers will walk |
03:13.21 | grimbeans | reliability issues huh :( |
03:13.34 | Hymie | well sure, if your bandwdith / isp is crap, that's not going to work well |
03:13.49 | Hymie | what did you plan for your bandwidth? |
03:14.04 | grimbeans | well, in the beginning I was going to start with just a 1.8 mbps cable line |
03:14.15 | Hymie | what's the upstream on that? |
03:14.16 | grimbeans | and add more cable bandwidth as necessary |
03:14.34 | grimbeans | hmm must be like 1/3 of it or so.. I gotta check.. :( |
03:14.48 | grimbeans | you're right, it's the upstream that matters :( |
03:15.06 | Hymie | doesn'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 |
03:15.35 | grimbeans | well, it's a "commercial" plan.. and the highest available here :( |
03:15.38 | Hymie | you might do better to find a GOOD isp, host a box there, and use a GOOD voip provider for incoming and outgoing |
03:16.05 | Hymie | the ISP will likely have free UPS service, as well as a generator with days of gas/diesel |
03:16.34 | Hymie | if 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 |
03:16.38 | Hymie | as long as you get a *good* isp |
03:16.41 | Hymie | with a good backbone |
03:16.47 | coppice | Uncontrolled Power Supply :-) |
03:16.52 | Hymie | heh |
03:17.32 | Hymie | grimbeans: 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 |
03:18.15 | killfill | hm.. 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. |
03:19.02 | Hymie | you could do that with polycoms.. what phones do you have? |
03:19.14 | Hymie | should be easy to do with any phone that has multiple presences |
03:19.41 | killfill | grandstream 2000 |
03:19.59 | killfill | it works when you call the phone directly (i.e. the sip extension) but not for the queue.. :S |
03:20.20 | killfill | http://pastebin.ca/629288 <-- thats my queue |
03:20.30 | Hymie | if you have multiple sip logins for those phones, you could just pretend they are multiple phones |
03:20.34 | Hymie | if you see what I mean |
03:21.13 | killfill | yes.. but i wish not to do that |
03:21.43 | killfill | i would need to double the config (users.conf), and every user will need to login/logout as agent several times per phone :) |
03:21.44 | Hymie | then 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..) |
03:22.44 | killfill | hm,.. |
03:23.09 | Hymie | killfill: this place is dead right now, it's usually hopping during the week |
03:23.18 | Hymie | killfill: if you're looking for more queue info, or more certainty on queues |
03:23.26 | Hymie | killfill: I'd try then... |
03:24.07 | killfill | try what?.. |
03:24.14 | Hymie | your questions |
03:24.32 | killfill | ah.. try then in the week.. :P |
03:24.37 | killfill | ok.. |
03:24.41 | Hymie | yeah |
03:24.50 | Hymie | I thinkmost people here are here on their work machines |
03:24.58 | grimbeans | hymie, 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.. |
03:25.11 | Hymie | grimbeans: such as? |
03:25.15 | grimbeans | let me explain my situation in a bit more detail.. :) |
03:25.18 | grimbeans | for one, |
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03:25.44 | grimbeans | I'm on an island in the pacific ocean, and the only close area is Guam |
03:25.53 | grimbeans | there are only 2 isps |
03:25.58 | grimbeans | the telco, and the cable company |
03:26.19 | Hymie | ok, your isp can be anywhere in the world |
03:26.21 | Hymie | keep that in mind |
03:26.23 | grimbeans | and I will be the only ITSP on this island when I start in the weeks to come |
03:26.24 | grimbeans | hmm |
03:26.44 | Hymie | so, what's more important is if someone already offers DNDs on your island |
03:26.55 | Hymie | check places like les.net to see ift hey do offer phone numbers there |
03:27.00 | Hymie | if they do, you can use them for incoming and outgoing |
03:27.09 | grimbeans | dnds? are they like dids? |
03:27.17 | Hymie | what country is this island? |
03:27.20 | Hymie | sure, dids I mean |
03:27.30 | Hymie | also, for example, les.net has 800 numbers |
03:27.40 | Hymie | which might be an option if you are outside of local did ranges |
03:27.47 | grimbeans | this is Saipan, of the Northern Mariana Islands, which is a protectorate of the U.S. (has a U.S. zip code, area code etc.) |
03:27.52 | grimbeans | hmm |
03:28.01 | Hymie | can you call US 800 numbers? |
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03:28.30 | Hymie | do you exist within the confines of thge US area code system? |
03:28.32 | grimbeans | some 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 |
03:28.42 | grimbeans | yes it has area code 670 :) |
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03:29.48 | grimbeans | hmm 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? |
03:30.12 | Hymie | no, you'd use their 800 did for incoming calls, then go out via their voip service |
03:30.18 | Hymie | from an ISP you setup on domestic US soil |
03:30.25 | Hymie | a big, reliable, cheap one |
03:30.43 | grimbeans | oh, ok.. I see.. |
03:30.45 | grimbeans | hmm |
03:31.12 | Hymie | sec |
03:31.24 | grimbeans | ok :) |
03:31.54 | Hymie | hmm |
03:31.54 | Hymie | yeah |
03:31.57 | grimbeans | okay I've actually thought of that earlier when I was brainstorming for options |
03:32.01 | Hymie | you'd have to do 800 for les.net it looks like |
03:32.13 | Hymie | that's your biggie, is 800 supported in your area |
03:32.17 | grimbeans | I opted not to go with it for a few reasons, although it would have been more stable |
03:32.19 | Hymie | beucase, this is waaaay cheaper for you |
03:32.23 | grimbeans | hmm |
03:32.32 | Hymie | it scales better too |
03:32.40 | Hymie | dynamically |
03:32.49 | Hymie | you don't have idle phone lines, or too few phone lines |
03:32.54 | Hymie | you have the precise right number of phone lines |
03:34.39 | grimbeans | yeah i'm here :) |
03:34.45 | grimbeans | i'm just thinking through what you've said |
03:35.05 | Hymie | that's a les.net 800 number |
03:35.12 | grimbeans | ok i'll try now :) |
03:35.33 | _DAW | I dont think you will find many ITSP's selling dids in the NM islands :) |
03:35.44 | Hymie | _DAW: there are thusands of them! |
03:35.55 | _DAW | really.. who? |
03:35.59 | Hymie | why |
03:36.01 | Hymie | that guy you know |
03:36.02 | Hymie | and |
03:36.07 | _DAW | just curious |
03:36.08 | Hymie | the other dude that hangs out with buddy |
03:36.09 | Hymie | you know |
03:36.12 | Hymie | the one with the face |
03:36.37 | _DAW | very nice.. I see you have it all lined up. |
03:36.51 | Hymie | hehe :Þ not sure about 800 number penetration though |
03:37.12 | grimbeans | I tried calling the number and it seems like it works.. |
03:37.17 | grimbeans | it gives me a short beep |
03:37.30 | Hymie | uh |
03:37.36 | Hymie | it didn't make it through to me ;) |
03:37.43 | grimbeans | which is unlike the recorded "this toll free number is not accessible from your service area" thingie |
03:37.44 | Hymie | so, it probably doesn't work :( |
03:37.45 | grimbeans | hmm |
03:37.48 | grimbeans | darn |
03:37.50 | grimbeans | :p |
03:37.51 | grimbeans | oh well |
03:38.07 | grimbeans | just confirmation of the |
03:38.17 | Hymie | but, emnail the les.net dudes! |
03:38.27 | Hymie | tell them what you want |
03:38.31 | grimbeans | poor connection that nmi have to the world :p :) |
03:38.32 | Hymie | they may say tough |
03:38.34 | Hymie | but you'll know |
03:38.36 | Hymie | also |
03:38.43 | Hymie | they do answer the phone almost immediately |
03:38.48 | Hymie | during normal hours |
03:38.52 | Hymie | so, you can try that too |
03:39.10 | grimbeans | ok I'll try it when it's work hours in the U.S. :) |
03:39.42 | grimbeans | now 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 |
03:40.23 | grimbeans | hmm.. maybe I need to think it through again lol.. |
03:40.28 | grimbeans | but here's what I thought |
03:41.01 | grimbeans | instead of the expensive T1 at $8,000 and only 24 voice lines, |
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03:41.13 | grimbeans | I would use regular analog lines at about $50 each |
03:41.32 | grimbeans | which comes out to $1,200 for 24 lines / mo |
03:41.34 | Qwell | who charged $8k for a T1? That's pretty ridiculous |
03:41.48 | Hymie | it's cheaper than your crazy telco charges for that t1 |
03:41.48 | Hymie | but |
03:41.54 | MACscr | Does anyone know of a outbound provider that allows for complete callerid rewriting? This includes the CNAM |
03:41.54 | grimbeans | yeah, it's RIDICULOUS but my telco has a monopoly on a lone island and there's no alternative |
03:41.56 | Hymie | there is also another issue ; installation coats |
03:41.58 | coppice | you should try to get that in the Guiness Book of Records - "Most expensive T1 in history" |
03:42.01 | Hymie | if you go to put in all those lines |
03:42.08 | Hymie | they're going to carge you big tiem for installation |
03:42.27 | grimbeans | :) |
03:42.28 | grimbeans | hmm |
03:42.32 | grimbeans | about the installation costs |
03:42.42 | grimbeans | I'm setting up shop in a major building |
03:42.44 | Hymie | no,you can't borrow barbra eden |
03:42.47 | grimbeans | that has |
03:42.58 | grimbeans | like over a hundred units |
03:43.04 | Hymie | good |
03:43.15 | Hymie | so you only have to write from the phone close to your server room |
03:43.18 | Hymie | wire |
03:43.26 | grimbeans | yep :) |
03:43.29 | Hymie | I think you'll be sorry with your cable though |
03:43.31 | Hymie | for bandwidth |
03:43.37 | grimbeans | T___T |
03:43.41 | grimbeans | it's either the cable co |
03:43.45 | grimbeans | or the telco |
03:43.53 | Hymie | I bet their cable isn't separate from their residential cable |
03:43.58 | Hymie | same for the telco |
03:43.58 | grimbeans | and the telco will hate me for taking their revenue and they suck anyways :( |
03:44.04 | Hymie | probably uses the same backbone, etc |
03:44.09 | grimbeans | yeah it's all together |
03:44.17 | grimbeans | same backbone :( |
03:44.25 | Hymie | if you can, you should *really* get into an ISP on the continent |
03:44.27 | Hymie | a good one |
03:44.41 | Hymie | if you can somehow get dids or 800s working locally |
03:44.45 | grimbeans | I guess I should think about that more seriously.. |
03:45.17 | grimbeans | I've considred DIDs but I don't think anyone is providing DID service to here |
03:45.32 | grimbeans | which seems logical since there is no ITSP here (except me starting up :P) |
03:45.38 | grimbeans | and as for the 800 service |
03:45.42 | grimbeans | some services may work |
03:45.44 | *** join/#asterisk andresmujica (n=andresmu@201.245.236.215) |
03:45.54 | grimbeans | but I think they charge me like $.02 per minute |
03:46.05 | grimbeans | which increases the rates I can offer to my customers :( |
03:46.19 | Hymie | well, you only need to undercut the telco |
03:46.20 | grimbeans | hmm |
03:46.29 | Hymie | and, international rates usually shine with voip |
03:47.06 | grimbeans | you're right.. :) |
03:47.22 | grimbeans | I probably should re-calculate costs before deciding |
03:47.25 | grimbeans | and |
03:47.34 | grimbeans | i should also call the telco on monday to check on the |
03:47.37 | coppice | we can't afford to use VoIP for international calls. We have to use the local telcos to save money :-) |
03:47.50 | grimbeans | ringdown service |
03:48.23 | grimbeans | but back to square one.. |
03:48.41 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
03:48.41 | grimbeans | if the 800s don't work out and if the telco refuses to provide ringdown service |
03:48.57 | grimbeans | I may need to rely on Automatic Call Distribution |
03:49.02 | grimbeans | to make it work at all |
03:49.16 | grimbeans | would it even be possible using Asterisk? |
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03:49.50 | Hymie | I've neerevern heard of that |
03:50.04 | grimbeans | T__T |
03:50.08 | grimbeans | well I've been reading for days |
03:50.10 | Hymie | I really think you're barking up the wrong trr |
03:50.14 | grimbeans | T__T |
03:50.17 | Hymie | how is it going to move the call to another line? |
03:50.20 | Hymie | call transfer? |
03:50.30 | Hymie | it has to tell the telco to move the call, after all |
03:50.39 | grimbeans | well that's what I thought |
03:50.53 | grimbeans | until I searched and searched and got some information about |
03:50.57 | grimbeans | ACD |
03:51.05 | shido6 | ? |
03:51.17 | grimbeans | and now I'm confused just as to what is possible with ACD |
03:51.33 | shido6 | 1 # ? |
03:51.36 | grimbeans | I'm wondering if it can only distribute WITHIN the network or IP |
03:51.38 | shido6 | 1 channel? |
03:51.55 | grimbeans | do you mean 1 analog channel? (i'm noob sorry :() |
03:52.09 | grimbeans | i plan to use 16 analog trunk lines initially |
03:52.12 | shido6 | is the 8xx number coming in voip or pri? |
03:52.15 | shido6 | or pots |
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03:52.37 | grimbeans | it's just a local access number with all regular analog POTS lines (no T1, etc.) |
03:52.45 | grimbeans | (T1 costs $8,000 here :() |
03:52.47 | shido6 | how many pots lines? |
03:52.49 | shido6 | good lord |
03:52.53 | grimbeans | haha yeah |
03:52.53 | shido6 | where are you ? Moon ? |
03:53.00 | Qwell | shido6: Saturn, I think |
03:53.03 | grimbeans | middle of the pacific ocean T________T |
03:53.04 | shido6 | in a volcano? |
03:53.16 | grimbeans | now it's a lifted limestone island but |
03:53.21 | shido6 | no SAT? |
03:53.25 | grimbeans | neighboring islands are volcanoes! that erupt! |
03:53.25 | shido6 | VSAT |
03:53.26 | shido6 | ? |
03:53.31 | grimbeans | VSAT? satellite? |
03:53.40 | shido6 | yeah |
03:54.07 | grimbeans | I'm not so familiar with satellite service (except satellite TV) |
03:54.16 | shido6 | google it |
03:54.18 | grimbeans | what is possible? there are some satellites that can be tapped here |
03:54.19 | shido6 | save your $8k |
03:54.22 | grimbeans | well |
03:54.26 | grimbeans | haha |
03:54.27 | shido6 | 250 ms |
03:54.42 | grimbeans | the thing is I need |
03:54.46 | *** join/#asterisk sysreq (n=sysreq@modemcable064.127-81-70.mc.videotron.ca) |
03:54.51 | grimbeans | to let people from my local area call me up |
03:54.57 | grimbeans | using PSTN |
03:54.59 | coppice | your calls go sailing on the Latent Sea |
03:55.16 | grimbeans | 250 ms really good for my location XD |
03:55.34 | grimbeans | it's like 200 ms - 500+ ms usually lol |
03:55.41 | grimbeans | even on wired broadband |
03:55.52 | shido6 | ... yeah |
03:56.00 | Qwell | grimbeans: what island? |
03:56.11 | grimbeans | saipan, of the northern mariana islands.. near guam :p |
03:56.37 | grimbeans | hmm |
03:57.05 | shido6 | tap into the fiber from the Commonwealth of Northern Mariana Islands Public School System |
03:57.08 | grimbeans | yeah anyways T1's $8K so i'm planning to just use regular analog lines, $50 each in conjunction with cable internet |
03:57.09 | shido6 | they will never know... |
03:57.11 | grimbeans | hmm |
03:57.12 | shido6 | (gulp) |
03:57.13 | grimbeans | hahaha |
03:57.45 | shido6 | actually |
03:57.49 | shido6 | you should ask. |
03:57.56 | JT | grimbeans: you should probably just give up |
03:58.02 | Qwell | JT++ |
03:58.03 | grimbeans | nooo T_T |
03:58.22 | grimbeans | well |
03:58.24 | grimbeans | ultimately |
03:58.25 | shido6 | they have a T1 to the internet |
03:58.34 | grimbeans | there's only one fiber optic cable |
03:58.39 | grimbeans | that connects this island to the world |
03:58.49 | shido6 | oh wow |
03:58.49 | grimbeans | and that cable is owned by the local telco |
03:58.52 | grimbeans | and |
03:58.54 | shido6 | how many channels did you need? |
03:58.56 | grimbeans | everyone else just rents it |
03:58.58 | shido6 | (simultaneous calls) |
03:59.10 | grimbeans | at first just 16 to keep costs low but |
03:59.21 | grimbeans | i expect I'd need like 50 to |
03:59.25 | shido6 | yeah |
03:59.28 | shido6 | thats gonna suck |
03:59.33 | grimbeans | possibly a few hundred eventually |
03:59.47 | Qwell | You're never going to get a few hundred analog lines... |
04:00.01 | shido6 | is there a GSM carrier over there? |
04:00.22 | grimbeans | the local telco just recently (a week or two ago) got GSM |
04:00.29 | grimbeans | nobody else has it :( |
04:00.49 | grimbeans | qwell, is that because of the installation costs? |
04:00.52 | shido6 | is the local telco an enemy today? |
04:01.02 | Qwell | No, they just aren't ever going to give you that many |
04:01.09 | Qwell | not when they're charging $8k for a T1 |
04:01.11 | grimbeans | not yet, but it will be very soon once I take their long distance and some local customers away |
04:01.16 | grimbeans | well |
04:01.20 | grimbeans | that's one thing though |
04:01.26 | grimbeans | they have a monopoly |
04:01.32 | grimbeans | they can't refuse service to anyone :D |
04:01.40 | Qwell | says who? :p |
04:01.51 | grimbeans | my lawyer anthony long that's who! :D |
04:01.58 | grimbeans | hehe |
04:02.02 | grimbeans | and in the mid '90s |
04:02.11 | shido6 | you're going to need "two-way" satelite internet |
04:02.15 | grimbeans | the telco got sued for antitrust violations once |
04:02.18 | grimbeans | hmm |
04:02.25 | grimbeans | shido, |
04:02.37 | Qwell | It would probably be cheaper to run a second fiber link to the island :p |
04:02.41 | grimbeans | would the satellite internet be any better than the cable internet i'd be getting? |
04:02.46 | grimbeans | except maybe stability |
04:02.50 | grimbeans | haha another fiber XD |
04:02.57 | coppice | Saipan, land of a million cheap tee-shirts :-) |
04:03.05 | grimbeans | it cost the telco like 16 million $ to connect saipan to Guam :D |
04:03.10 | coppice | This is the main fibre activity |
04:03.46 | shido6 | how many ppl will you service? |
04:03.59 | sevard | lulz, how many people will you service. |
04:04.01 | sevard | your mother. |
04:04.10 | grimbeans | well, the last company that tried something like this (and went under) |
04:04.26 | shido6 | fill out the form and call http://www.macrosat.com/macrosat-reseller.html |
04:04.27 | grimbeans | was using that $8,000 T1 line and they lasted maybe a bit short of a year |
04:04.32 | Qwell | gee, I wonder why they went under... |
04:04.42 | shido6 | bbl |
04:04.56 | grimbeans | actually it wasn't a revenue issue but some of the managers were diverting money.. :( |
04:05.30 | grimbeans | macrosat? is that the satellite internet service provider? |
04:05.36 | Hymie | at $300 a line it wasn't a revenue issue?! |
04:05.38 | JT | shido6: forget sat |
04:05.42 | JT | shido6: waste of time |
04:06.18 | coppice | satellite is the only way he will bypass the telco. |
04:06.25 | grimbeans | yeah they were doing quite well with the long distance calling service |
04:06.33 | Hymie | he could use a blimp ;) |
04:06.34 | grimbeans | often all lines would be busy |
04:06.42 | JT | bypassing the telco for resale would appear to be a waste of time |
04:06.43 | grimbeans | haha |
04:06.49 | grimbeans | i think |
04:06.56 | grimbeans | i gotta use the cable company for now |
04:07.04 | grimbeans | bad as that sounds :( |
04:07.07 | Hymie | FIND OUT ABOUT THE 800 NUMBERS |
04:07.07 | Qwell | JT: s/^.*\(resale\).*$/\1/ |
04:07.11 | JT | or just not do it |
04:07.12 | grimbeans | yup |
04:07.13 | grimbeans | i will |
04:07.15 | Qwell | erm |
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04:07.19 | grimbeans | i'll check the 800s |
04:07.20 | Qwell | JT: s/^.*\(resale.*\)$/\1/ |
04:07.43 | grimbeans | you mean resell voip service? :p |
04:07.57 | JT | i mean forget about it, seriously |
04:08.02 | JT | unless you have big dollars |
04:08.16 | grimbeans | well |
04:08.21 | Hymie | JT: he does, he used a photocopier to enlarge some ones |
04:08.39 | grimbeans | if i go the analog route.. here's what it would cost me initally with 16 lines |
04:08.43 | grimbeans | per month |
04:08.46 | grimbeans | i'd have to pay |
04:08.52 | grimbeans | $1,200 for pstn |
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04:08.56 | JT | does the telco run the cable? |
04:08.57 | grimbeans | $110 for cable |
04:09.18 | grimbeans | (they own the fiber optic cable but the cable company rents it and telco can't stop them) |
04:09.25 | grimbeans | $1,000 for rent |
04:09.28 | JT | what's the upstream on the cable? |
04:09.31 | grimbeans | $400 for utilities |
04:09.39 | grimbeans | I'm guessing 768 kbps :( |
04:09.42 | grimbeans | I may have to add |
04:09.46 | grimbeans | another line or two of cable |
04:09.53 | grimbeans | but that's pretty cheap so it's ok |
04:10.02 | grimbeans | and |
04:10.07 | killfill | Qwell: 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. |
04:10.32 | JT | grimbeans: what is the primary type of service you are offering? incoming DIDs, or outgoing calls? |
04:10.37 | killfill | eventually they could anwear the new call, putting the current one in hold. |
04:10.46 | grimbeans | call waiting service is available from the telco.. call forwarding on busy is not provided |
04:10.46 | coppice | grimbeans: you seem to be treating the internet bandwidth like its guaranteed. |
04:10.54 | grimbeans | and jt i'll be giving out |
04:10.57 | grimbeans | oops |
04:10.58 | killfill | (or anyone..) |
04:11.05 | JT | call waiting is completely useless to a pbx |
04:11.05 | grimbeans | i'll be providing three things |
04:11.58 | grimbeans | 1.) voip service to those who have broadband, 2.) prepaid long distance calling service, and 3.) local DID (considering cost currently) |
04:12.18 | JT | local did is the hardest |
04:12.26 | grimbeans | yeah i know :( |
04:12.28 | JT | without pri, it really is a waste of time |
04:12.31 | JT | or ss7 |
04:12.34 | grimbeans | T_T |
04:12.47 | grimbeans | i may have to just stick with the first two if the DID is not viable.. |
04:12.54 | Qwell | ~wglwat |
04:12.55 | jbot | wglwat is, like, well, good luck with all that |
04:13.00 | bkruse_home | Qwell++ |
04:13.08 | wunderkin- | wigglewat |
04:13.10 | grimbeans | hehe. thanks guys :* |
04:13.12 | Qwell | bkruse_home: wanna buy a T1 from Qwellcomm? |
04:13.21 | bkruse_home | Qwell: ill take 2 |
04:13.21 | Qwell | $7500 - that's a real bargain |
04:13.26 | wunderkin- | telecomjoshvoxmart? |
04:13.40 | bkruse_home | qwellcomm bought telecomjoshvoxmart out |
04:13.41 | Qwell | wunderkin-: Qwellcomm bought telcomjoshvoxmart |
04:13.50 | bkruse_home | thats old news. |
04:13.53 | wunderkin- | i missed that, will be at&t before you know it |
04:13.58 | grimbeans | xD |
04:14.07 | bkruse_home | we are working on it currently. |
04:14.18 | Qwell | Qwellcomm, the new AT&T |
04:14.28 | wunderkin- | sounds close enough to qualcomm |
04:14.30 | bkruse_home | i like it |
04:14.33 | Qwell | wunderkin-: yes |
04:14.50 | Qwell | and I'm gonna get sued for it one day, I'm well aware of that. |
04:14.56 | grimbeans | :D |
04:15.28 | grimbeans | (sorry to sound like a broken tape recorder but) |
04:15.40 | bkruse_home | -v grimbeans :P |
04:15.42 | grimbeans | so asterisk would not be able to let me do the following? |
04:16.00 | grimbeans | I'm going to be using 16 analog trunk lines initially by way of analog FXO gateways |
04:16.06 | grimbeans | the numbers will be 234-1001 through 234-1016 |
04:16.10 | grimbeans | I want all my customers to call just 234-1001 |
04:16.17 | JT | what sort of fxo gateways? |
04:16.17 | grimbeans | as each call is received, I want them to be automatically transferred (call distributed?) to one of the other 15 lines |
04:16.23 | grimbeans | so that the original 234-1001 is constantly free and open to new calls |
04:16.26 | grimbeans | hmm |
04:16.32 | grimbeans | they are audiocodes mediapack 108 |
04:16.44 | grimbeans | x 2 or x 3 as needed |
04:16.46 | JT | also, if you telco cannot do disconnect supervision you are also wasting your time |
04:17.10 | grimbeans | disconnect supervision? i'm not familar with it .. help :( |
04:17.24 | JT | polarity reverse on far end disconnect |
04:17.30 | JT | only needed for analogue lines |
04:17.35 | JT | because they have crap signalling |
04:17.47 | JT | so a computer/pbx can automatically detect the other end has disconnected |
04:17.58 | JT | otherwise you could be left with zombie lines for periods of time |
04:18.36 | grimbeans | hmm.. so to find out, I should ask the telco whether they have disconnect supervision capabilities? |
04:18.39 | Hymie | JT: 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.... |
04:18.57 | Hymie | JT: 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 ;) |
04:19.42 | JT | Hymie: i understand, sometimes the best option is to give up while you're ahead |
04:19.50 | grimbeans | noo XD |
04:19.51 | JT | grimbeans: yes |
04:19.55 | JT | find out |
04:19.55 | grimbeans | gyaa |
04:20.14 | grimbeans | ok i'll check on monday (their office is closed atm) |
04:20.16 | grimbeans | :( |
04:20.22 | grimbeans | so |
04:20.37 | grimbeans | if they have disconnect supervision capability, |
04:20.53 | grimbeans | asterisk would be able to do the automatic call distribution? |
04:21.02 | grimbeans | via my fxo gateway |
04:21.11 | JT | i have no idea what you mean via automatic call distribution |
04:21.24 | grimbeans | well supposedly |
04:21.39 | grimbeans | it can do all that call transferring / routing that I need |
04:21.48 | grimbeans | but I just am not sure I'm undestanding it right |
04:21.53 | grimbeans | for example |
04:21.55 | grimbeans | call centers |
04:21.58 | JT | i'm not sure what you mean |
04:22.10 | grimbeans | use Automated Call Distribution (ACD) systems to |
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04:22.29 | grimbeans | receive calls on one number and route them to various agents in their office |
04:22.29 | JT | doesn't that have more to do with agents? |
04:22.33 | grimbeans | yeah |
04:22.41 | grimbeans | but it can apparently also be used for voip |
04:22.46 | grimbeans | so basically |
04:22.48 | JT | not much to do with being an ITSP |
04:22.51 | grimbeans | my customer calls 234-1001 |
04:23.01 | grimbeans | and it automatically routes to some other of my 15 lines |
04:23.08 | grimbeans | and frees up 234-1001 for more incoming calls |
04:23.26 | JT | when your customer calls a number, your telco will put the call through to a line |
04:23.37 | grimbeans | well it's related to ITSP in the sense that |
04:23.47 | JT | if you have line hunt, it may put it through to the next available line in a group of lines |
04:23.50 | grimbeans | prepaid long distance calling will be accessible to customers by calling my number 234-1001 |
04:23.53 | grimbeans | hmm |
04:24.14 | grimbeans | line hunt / ringdown, as hymie and you mentioned earlier huh.. |
04:24.20 | grimbeans | another thing I should check with the telco.. |
04:24.32 | JT | and 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 |
04:24.38 | grimbeans | last time, I only asked about call forwarding on busy signal and they didn't have it |
04:24.42 | grimbeans | hmm |
04:24.48 | JT | forget large scale DIDs on analogue |
04:24.52 | JT | utter waste of time |
04:24.55 | grimbeans | ok :( |
04:24.58 | grimbeans | no DIDs then :( |
04:25.03 | JT | i mean |
04:25.11 | JT | you could have numbers for pre-paid calling |
04:25.12 | grimbeans | only voip service and prepaid long distance :( |
04:25.18 | JT | just can't have hundreds of numbers |
04:25.36 | grimbeans | hmm ok let me digest the above info .. :p :) |
04:26.47 | grimbeans | and |
04:27.02 | grimbeans | the reason why i can't have hundreds of numbers is.. |
04:27.08 | JT | you'll hate me, but where i am, i can get a 10 channel pri for about USD$175/mo :P |
04:27.08 | grimbeans | because the telco won't give it? |
04:27.18 | grimbeans | i hate you :( DIE! |
04:27.22 | grimbeans | lol |
04:27.32 | wunderkin- | yeah,me too |
04:27.35 | JT | because you have insufficient signalling to determine what number is being callED |
04:27.45 | grimbeans | but |
04:27.56 | grimbeans | wouldn't callerID do the trick? |
04:28.03 | grimbeans | (that's what i was counting on) |
04:28.23 | JT | that tells you who is callING |
04:28.32 | JT | unless your telco is willing to change that |
04:28.34 | [TK]D-Fender | grimbeans, You are dangerously clueless. Get a good lawyer |
04:28.43 | JT | to who is being callED, i doubt that they will |
04:28.45 | grimbeans | sorry TK i'm noob T_T |
04:28.53 | [TK]D-Fender | grimbeans, And some industrial strength antacid |
04:29.09 | grimbeans | :p |
04:31.08 | grimbeans | Thanks for all the advice, guys. I really appreciate it. :) |
04:42.51 | [TK]D-Fender | Crushing telco-wannabe's one newb at a time..... |
04:43.15 | coppice | [TK]D-Fender: you mean a dead lawyer? aren't they the only good kind? |
04:44.05 | [TK]D-Fender | coppice, no, living ones make better meat-shields when the ninja's come for you ;) |
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04:44.29 | coppice | I guess everything has some useful application |
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04:53.53 | evilicey | Anyone have any recommendations for a softphone ? |
04:56.30 | CunningPike | evilicey: SJPhone |
04:57.23 | Strom_M | CunningPike: i have a canada question for you :) |
04:57.45 | Strom_M | does there exist in calgary such a thing as a 24 hour drugstore, and if so, what would it be called? :) |
04:58.06 | coppice | a pusher? :-\ |
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05:01.14 | evilicey | thanks, i will give it a try :) |
05:01.19 | evilicey | also looking at X-lite |
05:04.15 | Juggie | Strom_M, you there still? |
05:04.26 | Strom_M | yes? |
05:04.31 | Juggie | shoppers drug mart |
05:04.40 | Juggie | will have a 24hr location somewhere |
05:04.45 | Strom_M | yes, i just let my fingers do the walking (tm) and discovered this |
05:05.00 | Juggie | your visiting calgary? |
05:05.16 | Strom_M | yes |
05:05.23 | Strom_M | s/your/you're/ |
05:05.49 | Strom_M | are you in 403 land? |
05:06.42 | Juggie | no |
05:06.44 | Juggie | i'm in ottawa |
05:06.59 | Juggie | also if you want to see if you can find something closer, you can also try rexall |
05:07.02 | Juggie | www.rexall.ca |
05:07.19 | Juggie | i just know the two big drug store chains are shoppers & rexall :) |
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05:08.32 | red9012 | how do online fax->email services work. which software are they using? |
05:09.31 | JT | there are probably many different setups |
05:10.09 | Strom_M | Juggie: yeah...shoppers drug mart is the only one which seems to go to the trouble of actually pointing out the 24 hour bit |
05:10.37 | Juggie | strom, yeah i just quickly looked @ rexall out of curiosity and it seems they are all closed by 10. |
05:10.58 | Juggie | the shoppers by me are all open until 12 midnight or later hence why i suggested |
05:11.35 | Juggie | it should only be like 11:15 in calgary now anyways so should be plenty open |
05:15.14 | Juggie | hope you find what you need :) |
05:15.18 | killfill | hm.. |
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05:36.35 | JT | all you north amercans |
05:36.38 | JT | americans |
05:36.49 | JT | if someone from overseas calls you, do you have to pay? |
05:36.56 | JT | if so, is it higher rates than normal? |
05:37.28 | grimbeans | no (as long as it's a landline), and no.. :p |
05:37.39 | grimbeans | erm you do mean PSTN right? |
05:37.57 | JT | yes |
05:38.07 | grimbeans | then no, and no.. :p |
05:38.15 | grimbeans | only they have to pay not you |
05:38.26 | JT | well in the US, if you're not on an unlimited plan, i know you need to pay to receive calls usually |
05:39.37 | grimbeans | hmm.. I'd personally never noticed charges for calls received, on the telco's phone bill, except for collect calls of course |
05:39.58 | grimbeans | they didn't even charge local calling rates when I received.. but that could just be a local thing where I was back then |
05:40.01 | DarKnesS_WolF | what is the best billing system for asterisk ? " webinterface "? |
05:40.23 | JackEStorm | on landlines no, unless you have a grandparents plan, or a DOD. |
05:41.27 | JT | JackEStorm: but on mobiles you do? |
05:42.29 | grimbeans | for mobiles it probably varies from region to region, provider to provider |
05:42.50 | grimbeans | on my little island they charge all calls received on mobile as though I were calling |
05:43.05 | grimbeans | but when I was back in the mainland they didn't charge for calls received on my mobile |
05:43.20 | grimbeans | so it probably varies |
05:43.26 | JT | heh, billing is such an utter mess over that way :P |
05:44.05 | grimbeans | :) |
05:44.48 | Juggie | JT, incomming calls are allways free. |
05:45.00 | JT | on mobiles apparently not |
05:45.07 | Juggie | except mobiles yes |
05:45.10 | Juggie | landlines are allways free |
05:45.28 | Juggie | but the rates dont change |
05:45.34 | Juggie | for mobile, they just use your minutes |
05:46.09 | Juggie | unless of course you are roaming |
05:46.11 | Juggie | but thats different |
05:47.59 | JT | hmm ok |
05:50.10 | Juggie | anyways, it makes no difference where the call orginated from |
05:50.18 | Juggie | you get charged the same |
05:50.59 | JT | ah ok |
05:51.11 | JT | the person i want to call is on nextel on generally only uses it for ptt |
05:51.19 | JT | he says his call rate is super high |
05:51.42 | Juggie | thats his plan though, not where the call comes from |
05:51.49 | JT | right |
05:52.07 | Juggie | any provider should offer unlimited evenings & weekends for cheap |
05:52.29 | JackEStorm | 'cept on burners |
05:52.42 | Juggie | burners? |
05:53.00 | JackEStorm | prepaid phones |
05:53.38 | CunningPike | Strom_M: Sorry - I was away for a bit |
05:54.53 | Juggie | JackEStorm, i guess.... plans are cheap |
05:55.05 | Juggie | i pay something retardely cheap, like 20$ + SAF |
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08:53.48 | Paul_UK | hi there, has anyone setup load balancing (not failover) with 2 asterisk servers? i.e using openser for example? |
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09:06.10 | Paul_UK | hmm, maybe no-one lol |
09:08.10 | tzafrir | Anyone probably hasn't set that up. |
09:08.24 | tzafrir | Haven't asked No-one yet. |
09:09.19 | JT | oh, you're finally around, tzafrir |
09:10.35 | JT | tzafrir: well, are you around? |
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09:17.17 | tzafrir | yes |
09:17.57 | tzafrir | JT, yes |
09:18.07 | JT | tzafrir: the A500 BRI drivers will use chan_woomera, i've been told |
09:18.45 | coppice | why are sangoma determined to do the dumbest things? :-\ |
09:19.03 | JT | you don't like it? |
09:19.50 | coppice | they 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 |
09:20.33 | JT | how far removed is the pri stack from bri? |
09:20.42 | coppice | its almost the same |
09:20.51 | Paul_UK | ok well thanks tzafrir |
09:20.58 | coppice | hence, bristuff is minor patches to libpri |
09:21.07 | tzafrir | minor? |
09:21.30 | JT | yeah, i thought it was weird considering there are plenty of pri stack |
09:21.35 | JT | going to woomera |
09:21.43 | coppice | yeah, pretty minor in the great scheme of things. most of it provides the card drivers |
09:22.56 | coppice | there are plenty of crappy PRI stacks. I don't know of one that could get through a compliance suite as it stands |
09:23.07 | coppice | a free one, that is |
09:23.24 | JT | heh |
09:24.45 | JT | going to write a good one? ;) |
09:25.24 | coppice | before the end of this year I intend to have one, unless something gets in the way |
09:26.03 | JT | feature wise will there be any improvements over what we already have? |
09:26.24 | coppice | it will pass compliance suites, for one |
09:27.10 | JT | that's one |
09:27.14 | coppice | In fact, it will actually come with one |
09:29.24 | coppice | that'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 |
09:30.07 | JT | stack compliance doesn't necessarily guarantee code stability though |
09:30.32 | coppice | stack non-compliance is almost a guarantee of trouble |
09:30.53 | JT | i don't doubt it |
09:31.02 | tzafrir | both chan_zap/bristuff and visdn claim to be "compliant", BTW. Not that I have bothered to verify it or check even to what standard |
09:32.07 | coppice | they claim to have gone through some approvals. different thing |
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10:17.53 | Paul_UK | Hey 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) |
10:32.29 | DragoraN | hi all! |
10:56.36 | JT | yawn |
10:57.13 | *** join/#asterisk kippi (n=none@untrust-gct.equinoxit.net) |
10:57.15 | kippi | hey |
10:57.31 | kippi | is there away to logon agents without being at there phone |
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11:06.52 | implicit | denon |
11:06.53 | implicit | wats up |
11:29.40 | DragoraN | lol |
11:30.41 | JT | hilarious |
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11:36.31 | hi365 | im 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" |
11:36.40 | hi365 | anyone familiar with this error? |
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11:43.18 | Paul_UK | so no-one knew an answer to my question. or am i gonna have to set it all up and see lol? |
11:43.18 | JT | the question was too big |
11:43.19 | JT | it also got chopped off |
11:43.38 | JT | also, i think you mean Mbit/s, not MB |
11:43.39 | Paul_UK | oh thanks |
11:43.48 | Paul_UK | 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). |
11:43.51 | Paul_UK | 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 centre has the priority when taking calls? Thanks |
11:43.57 | Paul_UK | hope thats better |
11:45.06 | JT | pretty sure they're not MB |
11:45.07 | JT | but umm |
11:45.12 | JT | load balancing might work |
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11:52.15 | Paul_UK | JT, they're not MB? |
11:52.24 | Paul_UK | oh Mbit? |
11:52.26 | Paul_UK | that better? |
11:53.05 | JT | yes |
11:53.24 | JT | i'm still not sure exactly what you're trying to do |
11:53.30 | JT | you have a few Internet links |
11:53.40 | JT | you have a main office and a call centre |
11:53.52 | JT | and 2 trunks, i don't know what type you're refering to there |
11:54.52 | Paul_UK | JT, ok alot is I dont know the terminology yet. |
11:55.11 | Paul_UK | so bear with me |
11:55.46 | Paul_UK | Ok, so with 2 main phone numbers, im assuming an inbound and outbound trunk would be associated with each? |
11:56.12 | JT | i have no idea how these phone numbers are delivered to you |
11:56.21 | Paul_UK | voip |
11:56.25 | Paul_UK | they arent pstn |
11:56.44 | JT | this company has no pstn? |
11:56.48 | JT | numbers are DIDs |
11:56.53 | Paul_UK | yeah |
11:57.50 | JT | no normal phone lines of some sort is a bad idea |
11:58.14 | Paul_UK | where i am currently, its total voip |
11:58.33 | Paul_UK | where i am moving too, the thought is to keep things the same |
11:59.28 | JT | really bad idea if that's VoIP over Internet |
12:01.15 | Paul_UK | well its an MPLS circuit to the providers network |
12:02.00 | jer | Paul_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) |
12:03.21 | Paul_UK | jer: I have just over a T1. 2MBit up and down, 1:1 contention |
12:03.46 | JT | Paul_UK: i don't think you're understanding, jer said "voice t1" |
12:03.54 | JT | not t1 Internet connection |
12:04.03 | Paul_UK | ok whats the diff? is it a leased line? |
12:04.12 | JT | it's a voice circuit |
12:04.21 | Paul_UK | ok well i've never heard of that |
12:04.34 | JT | most businesses with over half a dozen lines use a digital circuit |
12:04.40 | JT | usually PRI |
12:04.46 | JT | PRI can be T1 or E1 |
12:04.47 | jer | telephony terminology is fun to learn (and i say that as sarcastically as i can) |
12:04.50 | JT | in the UK it would be E1 |
12:05.06 | Paul_UK | well 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. |
12:05.19 | JT | some countries, including the UK have BRI available too |
12:05.24 | JT | basic rate interface |
12:05.24 | jer | Paul_UK, if it ends up being crap, you have a loss of revenue no doubt |
12:05.27 | jer | which is not a good thing |
12:05.29 | JT | as opposed to primary rate |
12:05.49 | JT | basic rate is 2 voice channels, and a signalling channel |
12:05.59 | Paul_UK | jer, 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. |
12:06.02 | JT | primary rate t1 is 23 voice channels and a signalling channel |
12:06.12 | JT | primary rate e1 is 30 voice channels and a signalling channel |
12:07.14 | Paul_UK | ok well i will talk with my provider on tuesday and see what they say |
12:07.52 | tzafrir | If you're from the UK, you have E1... |
12:08.24 | JT | mpls directly to provider is better than VoIPoI |
12:08.57 | Paul_UK | oh i was gonna say mpls |
12:09.10 | Paul_UK | its kinda like frame-relay? i think thats what we have |
12:09.16 | JT | no... |
12:09.37 | Paul_UK | ok well im wrong again lol.. but mpls is decent enough? |
12:09.47 | JT | that really depends who runs it |
12:10.05 | Paul_UK | wow so many variables, well this is my first implementation. if it fucks up, then it does. |
12:10.06 | JT | at least telcos with their PRIs are usually fairly likely to have reliability downpat :) |
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12:11.41 | Paul_UK | ok well with that out of the way, what about loadbalancing, will it fly? |
12:12.10 | JT | dunno |
12:12.17 | JT | are they bonded or just load balanced? |
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12:12.30 | Paul_UK | i would say load balanced |
12:12.58 | JT | if the load balancer operates correctly with respect to sip and rtp, then it could work |
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12:36.32 | jhiver_ | hi all |
12:36.59 | jhiver_ | 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 |
12:37.02 | jhiver_ | http://pastebin.ca/629552 |
12:37.57 | jhiver_ | i've tried to set core set debug and core set verbose but i don't see anything on the CLI |
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12:42.14 | DrukenLPY | morning everyone |
12:42.18 | jhiver_ | hi |
12:46.53 | jhiver | is 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 |
12:47.11 | DrukenLPY | pastebin it |
12:48.08 | jhiver | ok |
12:48.52 | jhiver | http://pastebin.ca/629561 |
12:50.18 | JT | which pastebins do you want us to look at? |
12:50.34 | jhiver | well the first is my sip.conf |
12:50.39 | jhiver | the second my extensions.conf |
12:51.19 | jhiver | i think the sip.conf is ok |
12:51.56 | JT | sorry, what are you using to call, what are you trying to call? |
12:52.00 | jhiver | sip debug says "Looking for 0017400062932847 in routes (domain 91.121.9.144)", and if leave just the following line: |
12:52.04 | jhiver | exten => 0017400XXXXXXXXX,1,Macro(balkana,${EXTEN:7}) |
12:52.14 | jhiver | then it doesn't match and i get a 404 |
12:52.18 | jhiver | ok |
12:52.31 | JT | your match is incorrect |
12:52.36 | JT | add a _ at the start |
12:52.36 | jhiver | I'm using a piece of software call voipswitch to place the call to the asterisk box |
12:52.44 | JT | i see |
12:52.49 | jhiver | 0017400062932847 |
12:52.53 | jhiver | 0017400XXXXXXXXX |
12:52.58 | JT | no. |
12:53.00 | jhiver | seems to match to me |
12:53.02 | JT | _0017400XXXXXXXXX |
12:53.06 | jhiver | aaaaah =) |
12:53.07 | JT | well you're mistaken |
12:53.09 | jhiver | silly me =) |
12:53.13 | jhiver | thx =) |
12:53.17 | JT | no underscore, no pattern match |
12:53.26 | jhiver | yes yes of course |
12:53.37 | jhiver | i'm all rusty with asterisk dialplans! thanks a bunch |
12:53.48 | JT | no probs |
12:54.28 | jhiver | is 'DBput' deprecated in 1.4 ? |
12:54.54 | jhiver | i have this now : No application 'DBput' for extension (macro-balkana, s, 3) |
12:56.58 | jhiver | aaah looks like they have become functions now? |
12:57.06 | jhiver | jeez, i'm real rusty =) |
12:59.32 | DrukenLPY | i'm having issues with realtime and odbc today... really annoying |
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13:22.42 | DragoraN | jhiver |
13:22.43 | DragoraN | Release 1.2 and later |
13:22.43 | DragoraN | DBPut(family/key=${foo}) (deprecated) |
13:22.43 | DragoraN | Set(DB(family/key)=${foo}) (new syntax) |
13:22.43 | DragoraN | Release 1.4 and later |
13:22.43 | DragoraN | Set(DB(family/key)=${foo}) |
13:22.51 | DragoraN | sorry for flood |
13:26.22 | jhiver | jeeez |
13:26.27 | jhiver | Random() is deprecated too? |
13:26.36 | jhiver | damn, the new syntax is ugly as hell =) |
13:26.51 | DragoraN | jhiver http://www.voip-info.org |
13:28.00 | DrukenLPY | jhiver: i agree with ya... it's very ugry |
13:28.21 | JT | it's more logical to use Set for everything than to have pointless little apps |
13:28.32 | JT | to set a variable... |
13:31.05 | DragoraN | i agree with JT :) |
13:31.28 | DragoraN | BFU's will never understand coding style.. |
13:31.30 | DrukenLPY | i agree as well, i just don't like the new syntax... |
13:31.42 | DrukenLPY | bfu's? |
13:32.25 | DragoraN | bfu - normal user :) |
13:32.31 | DragoraN | consumer ;) |
13:32.44 | DrukenLPY | how in hell does BFU stand for normal user or consumer? |
13:33.07 | DragoraN | bloody f... user |
13:33.22 | DrukenLPY | ahh, ok |
13:33.37 | DragoraN | an 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. |
13:33.57 | DragoraN | cz/sk only.. :) |
13:34.14 | DrukenLPY | well, that would make sence... since i'm canadian.. :) |
13:34.23 | DrukenLPY | EH! |
13:34.23 | jhiver | lads, is there an equivalent to dbdeltree on the CLI? |
13:36.19 | DragoraN | jhiver: dont be lazy :) |
13:36.25 | DragoraN | jhiver: database deltree |
13:36.34 | jhiver | let me see |
13:36.38 | DragoraN | Etch*CLI> database deltree |
13:36.38 | DragoraN | Usage: database deltree <family> [keytree] |
13:36.38 | DragoraN | <PROTECTED> |
13:36.38 | DragoraN | in the Asterisk database. |
13:37.17 | jhiver | cool |
13:37.48 | jhiver | so i can use it from the shell with the '-rx' flags... nice |
13:38.03 | DragoraN | awesome! :) |
13:38.43 | jhiver | sorry about this, i haven't touched asterisk since 1.0.9, it's changed quite a bit =) |
13:39.20 | DragoraN | jhiver: need a bridge course? |
13:39.40 | jhiver | heh, i think you've just made my day =) |
13:40.13 | DrukenLPY | 1.0.9 to 1.4.... you might as well forget everything you learned, and start over fresh :) |
13:41.51 | DragoraN | once upon a time, there was one voip |
13:43.04 | DragoraN | does anyone use iaxtermination ? |
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13:57.02 | DrukenLPY | well, looks like asterisk has been playing nice for about 2 hours |
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14:39.19 | jhiver | yo |
14:39.22 | jhiver | again =) |
14:40.35 | jhiver | is 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... |
14:42.12 | Strom_C | jhiver: asterisk is not a sip proxy |
14:42.42 | Strom_C | asterisk will put itself in the media path first and then only drop out if the other endpoints can talk directly |
14:42.52 | Strom_C | hence "back to back user agent" |
14:44.10 | kippi | is there away to logon agents without being at there phone |
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14:45.29 | Nugget | of course. just use addqueuemember |
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15:37.53 | hi365 | can i set a range in the dial plan? i currently have this : exten => s,n,Gosubif($["${CHANNEL:4:1}"="9"]?set1800|1) |
15:38.15 | hi365 | how can i make it "grater than 9 but less then 13"? |
15:38.30 | hi365 | of =9-13 |
15:38.36 | hi365 | of=or |
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15:51.24 | drumkilla | hm ... $[ $["${CHANNEL:4:1}" > "9"]$["${CHANNEL:4:1}" < "13"] = 11] |
15:51.50 | drumkilla | AEL would make that a whole lot easier, heh |
15:52.07 | drumkilla | and it would probably be most readable in conf format if you just break it up over a few lines |
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16:01.34 | hi365 | drumkilla: true but im planing on doing agent logins and its like 60 agents |
16:02.38 | hi365 | drumkilla: whats the "=11"? |
16:03.52 | drumkilla | well, it's two expressions, and then it's checking to see if both evaluated to 1 |
16:03.58 | drumkilla | and if so, the whole thing evaluates to 1 |
16:04.00 | drumkilla | (i think) |
16:09.35 | hi365 | drumkilla: ah! thanks ill try it |
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16:55.59 | Hmmhesays | oh I like vmware |
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17:26.36 | ukris | what is the default DTMF mode for Asterisk Voice Mail System .I am using asterisk 1.2.13 |
17:27.08 | Strom_C | DTMF is determined by the channel driver, not by the voicemail application |
17:28.26 | ukris | thanks Storm_C .I tried to channge the DTMF settiings on the sip.conf from RFC to INBAND .Dosent seem to work with the GXP2000 . |
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18:10.45 | hi365 | any enum guru's in the house? |
18:11.03 | hi365 | im getting error 504 when i try to register my sip url |
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18:26.36 | adderd | hello, I've got a quick question (hopefully) |
18:26.46 | adderd | Do either IAXTEL or DUNDi exist any longer? |
18:26.53 | adderd | both websites are missing in action |
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18:47.02 | adderd | is no one alive in here? |
18:48.35 | hi365 | not today |
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18:57.18 | adderd | does anyone know the where abouts of dundi or iaxtel? |
18:58.48 | *** join/#asterisk rmayorga (n=rmayorga@unaffiliated/rmayorga) |
19:18.47 | icanicant | i 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) |
19:22.35 | Gouroutrash | hello |
19:23.01 | x86 | icanicant: you can make Asterisk talk IAX2 to your ITSP, and only use a single port for signalling and RTP |
19:23.12 | x86 | if your ITSP doesn't suck |
19:23.36 | x86 | Gouroutrash: heya |
19:24.12 | Gouroutrash | anybody have a solution for kill all the beginners who use trixbox in the place of Asterisk ? |
19:24.14 | Gouroutrash | :) |
19:25.12 | icanicant | x86: 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. |
19:25.15 | Gouroutrash | i'm bored when i look questions on forums |
19:26.56 | *** join/#asterisk xuser (i=xuser@unaffiliated/xuser) |
19:27.28 | xuser | Hi, is there a open source billing software for VoIP Gateways? |
19:28.00 | *** join/#asterisk xrg_ (n=panos@88.218.82.82) |
19:28.35 | xrg_ | Juggie: hi.. |
19:28.46 | Gouroutrash | yes, "starshop" is opensource |
19:29.16 | Gouroutrash | i just know the name, never test it |
19:29.52 | xrg_ | Gouroutrash: I have: starshop is a minimal callshop implementation. |
19:30.11 | Gouroutrash | ok :) |
19:31.42 | xrg_ | 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) |
19:41.25 | DragoraN | what is better, asterisk or callweaver? |
19:41.37 | DragoraN | i like sqlite, but * is * |
19:50.18 | x86 | mysql > sqlite |
19:50.39 | x86 | callweaver has most of the features of asterisk, and more |
19:50.48 | x86 | like IAX2 / SIP jitterbuffers |
19:51.00 | x86 | and T.38 support |
19:54.36 | DragoraN | so callweaver? |
19:54.42 | DrukenLPY | anyone had a problem with database calls getting all screwed up after a while? |
19:55.54 | x86 | DrukenLPY: you mean CDR records stored in a database? |
19:56.00 | x86 | DrukenLPY: depends on what your needs are |
19:56.04 | x86 | err |
19:56.10 | x86 | DragoraN: depends on what your needs are |
19:56.31 | x86 | if you need generic jitterbuffer support in SIP and IAX2, and are doing T.38 faxing, use callweaver |
19:56.45 | x86 | if you want supportability and stability, get asterisk |
19:56.55 | DragoraN | x86: i want to use smart, extensible, opensource, well-documented, fast PBX with SIP, IAX, ... |
19:57.28 | x86 | well documented probably equates to asterisk |
19:57.59 | DrukenLPY | x86: nope... i mean odbc, and database gets... realtime, dialplan gets, everything |
20:02.04 | *** join/#asterisk op3r (n=Op3r@121.97.193.145) |
20:05.09 | x86 | DrukenLPY: never used ODBC for realtime, and wouldn't recommend doing so |
20:10.26 | jj56 | x86: why not? not using realtime, or just don't use it with ODBC? |
20:11.30 | jj56 | x86: i'm planning on using realtime with mysql, is that stable? |
20:11.35 | x86 | sure is |
20:12.27 | jj56 | how realtime is realtime.. do the updates to the dialplan happen during calls? or periodically? |
20:12.54 | x86 | all the time |
20:14.50 | jj56 | sounds good, do you know of any limitations such as maximum extensions / dialplan? |
20:15.27 | jj56 | i'm planning on having multiple customers design their own IVR using a webbased interface.. |
20:15.37 | jj56 | so the server would host 'multiple dialplans' |
20:16.08 | jj56 | that could result in a massive dialplan |
20:16.42 | jj56 | are there any limits to that? |
20:18.02 | DrukenLPY | x86: why no odbc for realtime? it's worked perfect for months.... and all of a sudden went screwy |
20:21.19 | adderd | anyone alive now? |
20:23.30 | jj56 | lemme check |
20:23.37 | jj56 | yeh :) |
20:25.03 | x86 | DrukenLPY: that's why i wouldnt recommend it |
20:25.18 | DrukenLPY | which is? |
20:25.37 | adderd | so whats the deal with DUNDi and IAXtel |
20:25.37 | x86 | ODBC |
20:25.40 | adderd | are they both dead? |
20:25.52 | x86 | adderd: IAXtel is run by someone at digium now |
20:26.00 | x86 | you have to email for an account, last i checked |
20:26.11 | x86 | mark gave me the email addy once but i lost it |
20:26.18 | adderd | well the website is dead anyway |
20:26.23 | x86 | yeah |
20:26.24 | adderd | same thing with Dundi |
20:26.48 | x86 | dundi is a dialplan function |
20:26.54 | x86 | "application" |
20:27.01 | adderd | the website for dundi is non existant now |
20:27.05 | adderd | that's what I'm saying |
20:27.11 | adderd | the directory is totally empty |
20:27.27 | adderd | makes 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) |
20:47.09 | kirberich | hi |
20:48.32 | cy- | sup kirberich |
20:48.42 | kirberich | i 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 |
20:49.12 | jj56 | still no solution eh? sorry to hear that.. |
20:49.36 | kirberich | jj56, yeah, i was about to throw the damn thing out of the windows yesterday ;) |
20:49.52 | jj56 | why don't you? |
20:50.09 | kirberich | but well, i know that it has to work somehow, i just need to find a person who's done it before ;) |
20:50.41 | kirberich | (getting it to work, not throwing it out the window) |
20:51.12 | jj56 | sorry, i threw a lot of stuff out of the window, but no hfc :) |
20:55.36 | jj56 | in 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) |
21:11.21 | gfgfgfgf | where does one start? |
21:12.20 | gfgfgfgf | can antone please help me i need to know what i need to get asterisk working? |
21:13.13 | DragoraN | gfgfgfgf: please tell us your problem first |
21:13.52 | DragoraN | gfgfgfgf: what do you need? x86 computer |
21:15.18 | gfgfgfgf | i 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!? |
21:15.24 | DragoraN | gfgfgfgf: http://www.voip-info.org/wiki/view/Asterisk+introduction |
21:16.07 | DragoraN | gfgfgfgf: try ubuntu |
21:17.16 | hypa7ia | gfgfgfgf: 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) |
21:18.27 | DragoraN | gfgfgfgf: http://www.voip-info.org/tiki-index.php?page=Asterisk+Linux+Ubuntu here you have cookbook for asterisk |
21:18.33 | DragoraN | gfgfgfgf: on ubuntu |
21:18.33 | gfgfgfgf | no what dependesies asterisk requires? |
21:18.44 | hypa7ia | gfgfgfgf: have a look at those links |
21:18.56 | DragoraN | try ubuntu.. dpkg will handle dependencies for you |
21:19.01 | hypa7ia | gfgfgfgf: what distro have you tried it on? |
21:19.11 | gfgfgfgf | suse 10.2 |
21:19.17 | hypa7ia | eep |
21:19.18 | hypa7ia | yeah |
21:19.31 | DragoraN | gfgfgfgf: ok.. you install asterisk using yast |
21:19.36 | hypa7ia | try ubuntu or CentOS, most people are on those |
21:19.45 | gfgfgfgf | no wget |
21:20.10 | hypa7ia | gfgfgfgf: do you have a reason you need to custom-compile it rather than using a package? |
21:20.38 | gfgfgfgf | yes want to use asterisk v1.4 |
21:20.54 | DragoraN | there are packages for 1.4 |
21:21.11 | gfgfgfgf | meaning? |
21:21.50 | hypa7ia | meaning you don't need to compile it |
21:22.06 | hypa7ia | you can use a package that someone else has compiled for you :) |
21:22.25 | hypa7ia | kind of like an msi on windows |
21:22.51 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
21:22.55 | DragoraN | gfgfgfgf: please, read briefly the at least first link i gave you... |
21:23.58 | gfgfgfgf | i read the handbook but cant test the dial plan unless asterisk starts |
21:25.00 | gfgfgfgf | where can i find a list of dependecies needed by asterisk |
21:27.27 | *** join/#asterisk Mahmoud (n=Mahmoud@unaffiliated/mahmoud) |
21:27.29 | Mahmoud | hello |
21:27.39 | Mahmoud | i'm using sipura 3000. how to skip * in my dialplan? |
21:27.50 | Mahmoud | not really related to asterisk, but i'm sure many people here use it |
21:28.07 | Mahmoud | i want to skip * and # |
21:33.47 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
21:33.57 | Strom_C | Mahmoud: what do you mean "skip * and #"? |
21:34.16 | Mahmoud | Strom_C: so i can dial a number that includes # |
21:34.24 | Mahmoud | sipura 3000 has a special meaning of # |
21:34.41 | Strom_C | Mahmoud: as does all of the rest of telephony |
21:34.46 | Strom_C | don't use # as part of a telephone number |
21:34.56 | Mahmoud | damn. it's already used by my ISP |
21:35.03 | Strom_C | # has meant "I am finished dialing; place the call now" since 1965 |
21:35.07 | Mahmoud | for special services, such as locking the phone down |
21:35.13 | Mahmoud | i know what # means |
21:35.23 | Mahmoud | i already skipped #, but forgot now |
21:35.47 | Mahmoud | problem is that, i can't dial special numbers that include #.. just because spa has special understanding |
21:36.13 | Strom_C | give me an example |
21:36.40 | hypa7ia | gfgfgfgf: if you install from a package it will load all the dependencies you need |
21:36.44 | hypa7ia | and then it will start |
21:36.50 | hypa7ia | give that a try :) |
21:37.20 | Mahmoud | Strom_C: for example, to lock my phone, i dial #33*1234# |
21:37.27 | Mahmoud | Strom_C: where 1234 is my pass key |
21:37.55 | *** join/#asterisk legis (i=legis@unaffiliated/legis) |
21:37.58 | hypa7ia | gfgfgfgf: do you know how to use YAST? |
21:38.34 | Strom_C | Mahmoud: that |
21:38.42 | Strom_C | that's kind of stupid |
21:38.49 | Strom_C | dumb ISP for the lose |
21:38.49 | Mahmoud | well, this is my ISP |
21:39.11 | Mahmoud | aghhh |
21:39.24 | Mahmoud | i already did it and it was working.. but today i removed it by mistake and saved the config.. damn it |
21:39.29 | legis | Hi, can * be use as a softswitch and billing VoIP platform? |
21:39.46 | Mahmoud | Strom_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) |
21:40.17 | hypa7ia | legis: as in for an ITSP? |
21:40.30 | *** join/#asterisk MrMister2 (n=mrmister@89-180-119-4.net.novis.pt) |
21:40.57 | legis | hypa7ia: yeah |
21:41.03 | legis | something like mera |
21:41.42 | hypa7ia | i don't know what mera is, but there are definitely people using asterisk as an ITSP platform |
21:41.47 | hypa7ia | but it'll take some config work |
21:42.47 | legis | basically what i want is to centralized various voip gateways through a softswitch. |
21:43.02 | legis | wondering if * is the a good tool for the job. |
21:43.31 | hypa7ia | centralize as in manage the billing for the other softswiches or as in replace them? :) |
21:43.59 | legis | replace |
21:44.46 | hypa7ia | i 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) |
21:45.43 | xuser | ok |
22:00.29 | Juggie | hypa7ia! |
22:00.43 | hypa7ia | Juggie! |
22:00.53 | hypa7ia | dude are you still in ottawa? |
22:01.02 | Juggie | yeah |
22:01.22 | hypa7ia | cool |
22:01.25 | hypa7ia | i'm there now |
22:01.33 | Juggie | i live in barrhaven now, bought a house |
22:01.35 | *** join/#asterisk easimon (n=easimon@baghira.kawo2.RWTH-Aachen.DE) |
22:01.42 | hypa7ia | coolio |
22:02.03 | Juggie | what are you doing back in ottawa |
22:03.10 | hypa7ia | some work for my dad |
22:03.41 | hypa7ia | you still at irs-north? |
22:03.42 | hypa7ia | :) |
22:05.09 | Juggie | yep |
22:05.17 | Juggie | still working for the feds in the market |
22:07.30 | Mahmoud | what are the cases when I get circuit busy ? |
22:07.45 | Juggie | well, when its busy. |
22:08.40 | Mahmoud | when the line is used by others? |
22:08.51 | Juggie | that would be one case yes |
22:08.58 | Mahmoud | another case is? |
22:09.41 | Juggie | well, it could be a group of lines thats entirely busy |
22:10.24 | Juggie | if your on a t1, its possible your telco could return a circut busy/congestion to you |
22:10.30 | Juggie | if they are unable to deliver your call |
22:10.31 | Juggie | etc |
22:13.00 | Mahmoud | using a single pots line |
22:13.04 | Mahmoud | and it tells me busy |
22:13.08 | Mahmoud | 100% sure no one uses the line |
22:13.23 | Mahmoud | will restart my SPA for final check.. |
22:13.44 | _DAW | Mahmoud: are you terminating the SPA on an * box? |
22:14.12 | Mahmoud | spa is connected to: |
22:14.13 | Mahmoud | analog phone |
22:14.13 | Mahmoud | asterisk |
22:14.13 | Mahmoud | pots phoneline |
22:20.27 | Mahmoud | damn |
22:20.33 | Mahmoud | no one using any line, and yet, congested |
22:22.42 | _DAW | are you getting congestion from the sipura directly? |
22:26.42 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
22:26.47 | Mahmoud | _DAW: -- SIP/pots-086f1000 is circuit-busy <---- SIP/pots is my sipura account |
22:28.24 | JT | sure you can probably try asterisk out via packages, some are better than others |
22:28.30 | JT | but generally you'll want to compile it |
22:29.38 | Mahmoud | it 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) |
22:39.11 | russellb | i'm just curious, what makes asterisk tough to deal with as a packaged app? |
22:39.29 | russellb | it's a common opinion, just hadn't spent much time thinking about why |
22:39.44 | Innatech | Usually the provided dialplans are somewhat labrynthine and hard to adjust if you want to do anything unanticipated. |
22:40.21 | russellb | well, i mean from an rpm, deb, or your package manager of choice |
22:40.27 | JT | russellb: they're always so old |
22:40.32 | russellb | not necessarily "packaged" as in with a gui .. |
22:40.34 | Innatech | ah. |
22:40.39 | russellb | JT: ah .. |
22:40.43 | JT | and sometimes you may wish to add patches before you compile |
22:41.07 | JT | and sometimes they come with on crack default configs |
22:41.14 | russellb | kind of like how many people are with their kernel? |
22:41.15 | JT | so it's a number of things i guess |
22:41.51 | russellb | fair enough, just curious. |
22:42.14 | DrukenLPY | russellb: it's a control issue :) |
22:42.33 | DrukenLPY | like how most drivers are horrible passengers.... |
22:42.52 | russellb | ha, nice |
22:43.31 | *** join/#asterisk techie (n=techie@c-67-181-184-170.hsd1.ca.comcast.net) |
22:44.00 | russellb | ManxPower: the Digium PBX is on 1.4 now :-p |
22:49.47 | CoolGuy21 | digium pbx? |
22:50.00 | CoolGuy21 | asterisk |
22:50.12 | JT | CoolGuy21: the digium pbx |
22:51.14 | DrukenLPY | digiums phone system |
22:52.31 | russellb | yeah, the main asterisk box at digium ... heh |
22:59.25 | Mahmoud | hmm |
23:06.25 | *** join/#asterisk mtaht4 (n=m@cpe-74-76-23-86.nycap.res.rr.com) |
23:09.45 | Mahmoud | Strom_C: spa says that the pots line is idle.. how come it says its busy? weird :/ |
23:10.35 | Mahmoud | running asterisk -rvvvvvvvvvvvvvvvvvvv |
23:10.50 | Mahmoud | when I call any phone in the pots network, i get this error: http://pastebin.ca/630156 |
23:11.46 | Mahmoud | SIP/pots is SPA's account regarding its POTS interface |
23:12.17 | Mahmoud | the interface connected to the ISP |
23:13.39 | JT | pretty useless cli output |
23:13.44 | JT | at least turn on sip debug |
23:13.51 | JT | and does the spa have its own log? |
23:14.08 | Mahmoud | nope |
23:14.15 | Mahmoud | couldn't find spa logs |
23:19.37 | *** join/#asterisk hohum (n=dcorbe@dhcp64-134-231-245.shs.nyc.wayport.net) |
23:21.24 | Mahmoud | JT, does this say anything usefull? http://pastebin.ca/630164 |
23:21.44 | Mahmoud | jt, asterisk is 10.1.0.1 and spa device is 10.1.0.3 |
23:21.59 | JT | cope |
23:22.01 | JT | nopw |
23:22.25 | JT | why would you paste just one message in the sip dialogue? what a waste of time |
23:22.30 | JT | full call or don't bother |
23:23.05 | Mahmoud | any easy way to select the whole text? |
23:23.13 | Mahmoud | scrolling down and copy pasting is a bit tough |
23:23.49 | JT | you could extract it from the full log if it's any easier |
23:24.09 | Mahmoud | where is it located? |
23:24.34 | JT | if it's enabled, /var/log/asterisk/full |
23:27.01 | Mahmoud | how to enable it? :/ |
23:27.07 | Mahmoud | nvm |
23:30.57 | DrukenLPY | http://www.pastebin.ca/630171 |
23:32.31 | Mahmoud | JT, http://pastebin.ca/630172 |
23:35.29 | Mahmoud | hmmm |
23:39.13 | Mahmoud | JT, i think you already answered a similar question in mail-archive.com |
23:39.27 | Mahmoud | JT, 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) |
23:41.24 | puzzled | evening all |
23:42.14 | Juggie | puzzled |
23:42.36 | JT | Mahmoud: that's not me |
23:42.47 | Mahmoud | JT, i see |
23:43.00 | Mahmoud | jt, i disabled qualify lines in sip.conf and still doesn't work.. |
23:45.24 | JT | Mahmoud: you did not paste some important stuff from later in the sip dialogue |
23:45.37 | JT | i really do mean paste everything |
23:45.46 | JT | not what you think is important |
23:46.04 | JT | everything from just before you make the call, to just after it fails on the asterisk console |
23:46.14 | Mahmoud | ok |
23:46.18 | JT | otherwise you are wasting time |
23:49.24 | Mahmoud | JT, full log http://pastebin.ca/630198 |
23:49.46 | Mahmoud | JT, the starts by starting asterisk till the call is end |
23:50.52 | *** join/#asterisk |dennis| (n=dennis@200.32.236.20) |
23:53.12 | JT | the spa probably doesn't know how to route that call |
23:53.32 | JT | also, looks like you're misusing qualify= in your configs, as there's tonnes of errors about them |
23:55.00 | Mahmoud | JT, hmm weird.. will check if i was editing wrong file or no |
23:55.04 | Mahmoud | s/no/not/ |
23:57.11 | Mahmoud | JT.. ohh sorry, wrong log file |
23:59.19 | Mahmoud | JT, currect log file http://pastebin.ca/630211 |
23:59.29 | DrukenLPY | can someone help me with a realtime problem ? |
23:59.40 | Mahmoud | JT the previous log file wasn't updated.. it was a copy of a previous one |