| 00:01.37 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 00:03.41 | Kyosh | recommended? |
| 00:03.42 | Kyosh | heheh |
| 00:03.43 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 00:03.53 | Kyosh | i know im using a bunch cause none are too reliable |
| 00:03.59 | infinity1 | lol |
| 00:04.04 | infinity1 | which are you using? |
| 00:04.18 | Kyosh | for orig or term? |
| 00:04.22 | Kyosh | dom or int? |
| 00:04.22 | infinity1 | both |
| 00:04.30 | Kyosh | dont get snippy with me kiddo |
| 00:04.39 | infinity1 | orig and term for domestic US |
| 00:05.12 | jaytee | ~itsplist-us |
| 00:05.13 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 00:05.18 | Kyosh | globalpops for orig, some term, lcr360 for term (they suck actually i found out, laundry list of problems), was using vitelity but they got stupid with me |
| 00:05.22 | jaytee | ~itsplist-uk |
| 00:05.23 | jbot | extra, extra, read all about it, itsplist-uk is UK based ITSps include http://www.voiptalk.org/ http://www.voipon.co.uk/ http://www.gradwell.com/ and a few other tinpot companies you can dig up with google. |
| 00:05.37 | jaytee | ~itsplist-eu |
| 00:05.41 | Kyosh | bandwidth.com is a joke. they only resell other providers |
| 00:05.58 | Kyosh | broadvoice, voicepulse, they are ok, but they dont do wholesale |
| 00:06.20 | lesouvage | hello, I have a weird problem. I have a script that generates callfiles using a list of telephne numbers. If I use this script from the Asterisk cli with the proper parameters copied from the Asterisk cli it is working ok. When I launch the script using the System() app in the dialplan Asterisk is complaining about invalid file content but is launching the call three times. Any suggestion or... |
| 00:06.21 | lesouvage | ...pointers? |
| 00:06.34 | Kyosh | tsg is good but they require so damn much money minimum every month |
| 00:06.53 | lesouvage | asterisk cli=linux cli |
| 00:07.21 | jaytee | aha! so you didn't really launch the script from the Asterisk CLI!!! No soup for you!!!! |
| 00:07.27 | lesouvage | the first asterisk cli (sorry, back in full typo mode) |
| 00:07.29 | Kyosh | heheh |
| 00:08.02 | jaytee | Kyosh, what about Level3? I think they do wholesale voip |
| 00:08.34 | Kyosh | they do but require a very large volume |
| 00:08.47 | Kyosh | otherwise you have to buy from a reseller |
| 00:08.52 | Kyosh | then there is commpartners |
| 00:08.57 | Kyosh | havent used them in a while |
| 00:08.59 | Kyosh | UTI was a joke |
| 00:09.07 | lesouvage | jaytee: The script has different results depending on how it was lounched. When using the System() app within the dialplan things go wrong, when launched (just for testing) from the linux prompt it works ok. |
| 00:09.30 | infinity1 | Kyosh: is tsg 4tsg.com ? |
| 00:10.56 | Kyosh | nopes |
| 00:11.09 | jaytee | lesouvage, my psychic powers tell me that something is wrong in the way it's called in the dialplan but I'm fresh out of tea leaves and chicken entrails so I can't be more precise. |
| 00:11.13 | Kyosh | tsg is a wholesale buyer from level3 and others but they have great prices |
| 00:11.27 | infinity1 | Kyosh: whats their website? |
| 00:12.06 | Kyosh | tsgglobal |
| 00:12.08 | Kyosh | sukie site |
| 00:12.23 | lesouvage | jaytee: if I pastbin it somewhere will you take a quick look? |
| 00:12.25 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 00:13.29 | jaytee | lesouvage, yeah |
| 00:14.35 | *** join/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 00:14.35 | *** mode/#asterisk [+o Mog] by ChanServ |
| 00:18.44 | *** join/#asterisk digitalirony (i=digitali@my.grandma.uses.shellium.org) |
| 00:20.01 | lesouvage | jaytee: This is it http://www.pastebin.be/17391 |
| 00:21.28 | *** part/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 00:23.03 | jaytee | lesouvage, sorry, nothing's coming to me |
| 00:25.07 | *** join/#asterisk RichiH (i=richih@freenode/staff/richih) |
| 00:26.10 | jaytee | wow, never saw a bot suckup like that before |
| 00:27.41 | lesouvage | jaytee: I think he is addressing you ;-) |
| 00:28.27 | jaytee | no, he was addressing RichiH from freenode staff. |
| 00:29.04 | lesouvage | jaytee: Do you see anything strange in the callfiles (except that the system added some lines while generating calls in the process) |
| 00:29.46 | lesouvage | http://www.pastebin.be/17392 |
| 00:30.02 | jsmith | ~onjoin RichiH |
| 00:30.18 | jsmith | ~yow |
| 00:30.34 | jaytee | hahaha |
| 00:30.56 | jsmith | ~buy jaytee donuts |
| 00:30.57 | jbot | ACTION goes to S-Mart and gets a dozen donutss for jaytee |
| 00:31.08 | jaytee | ~botsnack |
| 00:31.08 | jbot | jaytee: :) |
| 00:31.18 | jaytee | I'm the only one around here that ever feeds him |
| 00:31.23 | jaytee | poor bot! |
| 00:31.23 | jsmith | ~factinfo jaytee |
| 00:31.23 | jbot | there's no such factoid as jaytee, jsmith |
| 00:31.55 | jsmith | jbot: jaytee is likes to play with the animals in the zoo |
| 00:31.55 | jbot | jsmith: okay |
| 00:31.59 | jsmith | ~jaytee |
| 00:32.00 | jbot | somebody said jaytee was likes to play with the animals in the zoo |
| 00:32.45 | jaytee | any vodka in that mango smoothee? |
| 00:35.46 | Kyosh | rum rum rum |
| 00:38.23 | jaytee | lesouvage, I'm not that well versed in callfiles. I don't see anything obviously wrong in the pastebin but since some of the vars are using what I assume is Danish or some other language I'm having problems following the logic. |
| 00:39.10 | Chainsaw | jaytee: It's flemish or dutch. |
| 00:39.10 | lesouvage | Some variables are named in Dutch |
| 00:39.17 | Chainsaw | jaytee: Betting on dutch because of the 31 country code. |
| 00:39.32 | Chainsaw | jaytee: BELLEND_NUMMER is 'incoming number', KLANT is customer. |
| 00:39.59 | jaytee | ok, dutch it is then. I still don't see anything in the code that would make it try to loop or run through 3 iterations |
| 00:43.20 | lesouvage | I think I know the answer. The context is also called by a callfile but there is no Answer() at the top so no signal is giving back that the local channel is actualy "picked up". Now I added an Answer() at the beginning and now I just have one call. I can go to bed ;-) |
| 00:44.01 | jaytee | cool! |
| 00:44.07 | lesouvage | jaytee: thanks for the effort. |
| 00:44.26 | jaytee | lesouvage, your welcome. |
| 00:44.47 | jaytee | must be late there, get some rest |
| 00:44.48 | Chainsaw | lesouvage: Weltrusten. |
| 00:44.56 | lesouvage | see you |
| 00:45.07 | Chainsaw | jaytee: (01:47 local time there) |
| 00:45.10 | jaytee | Chainsaw, what's that mean in Englsih? |
| 00:45.15 | Chainsaw | jaytee: Good night. |
| 00:45.16 | jaytee | or English even |
| 00:45.46 | jaytee | ah, cool! is the W pronounced like a v? |
| 00:46.20 | Chainsaw | No, like a regular w actually. You're thinking of polish :) |
| 00:46.41 | jaytee | ok |
| 00:48.36 | jaytee | yeah, I've made alot of friends over the years who are of polish ancestry. One of my friend's last name is Musewicz and it's pronounced muh-seh-vitch |
| 00:49.55 | Chainsaw | *nod* |
| 00:52.56 | drmessano | So 3 of Jaytees friends get on an airplane with a priest |
| 00:53.03 | drmessano | Ba-dum-ching |
| 00:53.35 | jaytee | don't quit your day job |
| 00:53.56 | drmessano | and please, try the fish |
| 00:54.19 | drmessano | THANK YOU, GOODNIGHT |
| 00:57.37 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
| 01:02.57 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
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| 01:19.41 | k-man | what are "queues" in asterisk? |
| 01:26.09 | k-man | [TK]D-Fender: you awake? |
| 01:26.37 | drmessano | Hes not |
| 01:26.42 | k-man | oh |
| 01:27.15 | k-man | when moh is not required, it seems asterisk suspends the moh process |
| 01:27.32 | k-man | is there a way to make asterisk kill the process when not required rather than suspend it? |
| 01:28.20 | drmessano | What "process"? Its a module |
| 01:28.32 | drmessano | Why would you need to kill it? |
| 01:29.11 | jaytee | queues are for answering multiple calls to the same number and having the calls directed to queue "members" in whatever manner you choose, i.e. whichever member hasn't taken a call in the longest time period or ringing all available member phones at once. |
| 01:29.44 | k-man | drmessano: oh - a custom one, it calls mplayer |
| 01:30.14 | k-man | jaytee: ah, thanks - so it can be used in say, call centers? |
| 01:30.39 | jaytee | k-man, yes it's usually exactly where it's used in most situations. |
| 01:31.33 | jaytee | or if you have one main DID number pointing to a "receptionist" or an IVR and you want to control the flow of the calls, throttle the number of calls entering the IVR for some purpose, etc. |
| 01:31.35 | *** join/#asterisk HouseMD (n=nandateb@unaffiliated/geek) |
| 01:31.37 | k-man | drmessano: i need to kill it because i have set up streaming audi of a radio station as moh. the problem is, once moh is not required, it suspends the process. mplayer has a cache so when asterisk resumes the mplayer process, you hear about 10 seconds of audio, which is old and no longer live, then it stops for a few seconds and then resumes with fresh audio |
| 01:31.54 | k-man | jaytee: i see, thanks |
| 01:32.26 | k-man | drmessano: i'd like to eliminate that problem |
| 01:32.40 | drmessano | I dont get into the whole streaming-audio-as-moh thing, since 99% of the time, people are not doing it legally |
| 01:33.03 | jaytee | some serious fines if you get caught |
| 01:33.22 | k-man | drmessano: oh - well, actually, its just for my home use - and i don't even play it to people on hold. i just have it so i can dial a number and listen to the radio |
| 01:33.38 | drmessano | Tell that to the judge |
| 01:33.45 | drmessano | Slimeball |
| 01:33.47 | k-man | its kinda cool too - cos i dial 702, and get the radio station on AM 702khz |
| 01:33.56 | drmessano | heh |
| 01:33.58 | drmessano | Oh...... |
| 01:34.02 | k-man | or say, dial 630 and get that "frequency" |
| 01:34.03 | drmessano | You said the magic words |
| 01:34.05 | drmessano | AM |
| 01:34.13 | k-man | good old am |
| 01:34.19 | drmessano | I kid, I kid |
| 01:34.36 | drmessano | AM can kiss my ass |
| 01:34.38 | k-man | well, in sydney, they stations i like happen to be on AM. however in other parts of the country (in australia) they transmit them in FM |
| 01:35.06 | k-man | but technicaly speaking, i'm not listening to AM, i am listening to some sort of TDMA |
| 01:35.16 | k-man | sorry, TDM |
| 01:35.32 | drmessano | AM Radio is 1% of the revenue for a group of radio stations, and 99% of the work |
| 01:35.54 | k-man | drmessano: dunno - the stations i listen to are government owned |
| 01:36.01 | k-man | anyway - any idea how to do what I want to do? |
| 01:36.11 | drmessano | Dynamite |
| 01:36.15 | drmessano | No, I dunno |
| 01:36.20 | k-man | ok, mv |
| 01:36.24 | k-man | never mind i mean |
| 01:36.32 | drmessano | I didnt realize I was talking to someone in OZ |
| 01:36.43 | k-man | yes - you are :) |
| 01:36.52 | drmessano | I didnt think you guys could get a signal out from there now, with that firewall and all |
| 01:37.05 | k-man | drmessano: yeah - tell me about it |
| 01:37.16 | k-man | i don't quite know what the gov. has been smoking |
| 01:37.16 | drmessano | So let me get this right |
| 01:38.05 | drmessano | The govt there blocks legal and illegal web surfing, words with the letter Q, Z, and P everywhere else, but not IRC? |
| 01:38.26 | k-man | drmessano: actually, they aren't blocking anything yet |
| 01:38.34 | k-man | that legislation was shot down afaik |
| 01:38.34 | drmessano | Can I get a fair dinkum? |
| 01:38.52 | k-man | only if you have a you beauty |
| 01:40.05 | drmessano | Funny thing is |
| 01:40.42 | drmessano | How the hell would Australian ISPs even implement a govt mandated firewall? They cant even implement SERVICE |
| 01:42.18 | Pan3D | lol |
| 01:42.54 | drmessano | I remember freakin DingoBlue internet |
| 01:43.46 | drmessano | "Dingo ate my connection" was a running joke |
| 01:44.33 | Pan3D | oh this is interesting... |
| 01:44.34 | Pan3D | http://www.google.com/hostednews/ap/article/ALeqM5ibxVSu1tEp5K-rd29EHncvz6OggAD971RA806 |
| 01:44.40 | Pan3D | Sorry, a bit off-topic, but still |
| 01:45.12 | Pan3D | looks like it's still moving forward :/ |
| 01:45.46 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 01:46.31 | drmessano | http://www.theregister.co.uk/2009/03/02/oz_firewall_finished/ |
| 01:50.13 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 01:50.45 | Pan3D | I'm confused. There are a bunch of articles dated within the past two weeks discussing it. |
| 01:51.36 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
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| 01:58.53 | k-man | so im trying to make a dial plan to record a sound, then play it back - i got this from the wiki but its not quite working |
| 01:58.56 | k-man | http://pastebin.ca/1366753 |
| 01:59.04 | k-man | it seems to record the sound fine, but cannot play it back |
| 02:00.14 | k-man | i included the errors in the paste btw |
| 02:00.19 | *** join/#asterisk Magicblaze007 (n=sony@fl-67-233-204-182.dhcp.embarqhsd.net) |
| 02:02.03 | Pan3D | k-man: look at the paths |
| 02:02.36 | Pan3D | <PROTECTED> |
| 02:02.47 | Pan3D | <PROTECTED> |
| 02:03.06 | k-man | Pan3D: i think i found the problem, it was a . in the EXTEN part instead of a : |
| 02:03.42 | Pan3D | yep |
| 02:03.59 | Pan3D | it's the little things |
| 02:05.16 | k-man | yep, the really little things |
| 02:09.11 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
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| 02:36.20 | k-man | how can i normalise a bunch of wav files so they are all normalised relative to each other? |
| 02:41.31 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 02:46.37 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 02:48.26 | *** join/#asterisk dandate2 (n=dandate2@173.11.82.122) |
| 02:48.45 | dandate2 | i learned that my docsis business class cable won't handle the call volume i'm getting using alaw/ulaw, what should i do?? |
| 02:49.15 | drmessano | It wont? |
| 02:49.26 | dandate2 | nope it gets choppy at peak hours |
| 02:49.41 | dandate2 | someone else told me i would need T1 voice |
| 02:49.52 | drmessano | How many concurrent calls? |
| 02:50.10 | dandate2 | someone else told me to just use GSM, but my friend told me no |
| 02:50.18 | dandate2 | well that would be 8 agents on and 10 calls coming in |
| 02:50.33 | dandate2 | so roughly 20 channels being used |
| 02:51.09 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 02:51.17 | dandate2 | my friend says "You don't NEED GSM just a compression scheme you can use with our provider" |
| 02:51.27 | drmessano | Sounds like youre just maxing out the bandwidth |
| 02:51.39 | dandate2 | but my provider just told they only support alaw/ulaw lol |
| 02:51.46 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 02:52.02 | lanning | move to another provider? |
| 02:52.05 | dandate2 | yes bandwidth was maxed out |
| 02:52.13 | dandate2 | even the local network was slowing down |
| 02:52.44 | drmessano | Change providers |
| 02:52.53 | drmessano | or get more bandwidth |
| 02:53.02 | dandate2 | i got the most premium class from comcast |
| 02:53.16 | dandate2 | can anyone recommend a good DID provider that will offer compression options? |
| 02:53.36 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-191-112.lns10.mel4.internode.on.net) |
| 02:53.36 | drmessano | "compression" is some bullshit your dumbass friend made up |
| 02:53.40 | dandate2 | lol |
| 02:53.53 | drmessano | You need someone that supports less costly codecs |
| 02:54.27 | dandate2 | and that'l cost more no? i'ma ready to pay i been going with $8/mo -per did |
| 02:54.32 | drmessano | Buy some G729 licenses |
| 02:54.42 | drmessano | Get someone who supports G729 |
| 02:54.44 | drmessano | Be done with it |
| 02:54.47 | dandate2 | alright |
| 02:54.55 | dandate2 | how much will the G729 license run me? |
| 02:55.20 | drmessano | $10 per pair of channels |
| 02:56.26 | dandate2 | so roughly $200 for 20 channels? |
| 02:56.33 | dandate2 | who is that payable to anyway? lol |
| 02:56.37 | drmessano | Digium |
| 02:57.08 | dandate2 | alright thats not a bad deal then someone else told me it would cost me per minute or something i thought they were pulling my leg |
| 02:57.10 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:57.41 | drmessano | You have some stupid friends |
| 02:57.49 | drmessano | Seriously |
| 02:57.56 | drmessano | They tell you some effed up crap |
| 02:58.31 | drmessano | Bunch of effing skype or vonage users that think they're "VoIP guys" |
| 02:58.35 | drmessano | Tell them to go back to Yahoo |
| 02:59.08 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 03:00.14 | lanning | hey now, you have a Yahoo employee in here now... |
| 03:00.17 | lanning | :P |
| 03:01.31 | drmessano | Yahoo is for soccer moms and old people |
| 03:01.42 | dandate2 | your right drmessano i am so sick of the "Experts" telling me conflicting shit |
| 03:01.48 | dandate2 | especially cuz i'm a noob this is hella traumatizing |
| 03:05.48 | k-man | where should i pust some custom sounds i made that I want to use as prompts in asterisk? |
| 03:05.55 | k-man | what path i mean? |
| 03:05.59 | k-man | so asterisk can find them |
| 03:06.35 | lanning | with the rest of the sounds? (/var/lib/asterisk/sounds) |
| 03:06.42 | k-man | i put them in /usr/share/asteris/sounds/custom but asterisk doesn't seem to find them |
| 03:06.55 | k-man | for some reason, on debian, all the sound files are in /usr/share/asterisk/sounds |
| 03:07.30 | k-man | i put them in a directory called custom in /usr/share/asterisk/sounds/custom |
| 03:07.33 | lanning | are you using "custome/filename" (without the extension) in your dialplan? |
| 03:07.44 | k-man | lanning, yes |
| 03:07.57 | lanning | what format are the sounds in? |
| 03:08.14 | k-man | wav |
| 03:08.30 | lanning | try convertion them to gsm |
| 03:08.38 | lanning | *converting |
| 03:08.50 | k-man | ok |
| 03:11.50 | k-man | ah - for some reason audacity saved it as 44.1khz |
| 03:11.58 | k-man | converted it back to 8khz and it works |
| 03:12.14 | hardwire | that happens |
| 03:14.56 | k-man | it could have just been my inability to use audacity, but i found it tends to stick periods of silence onto the ends of files |
| 03:15.12 | [TK]D-Fender | I'll believe that :) |
| 03:18.44 | k-man | morning [TK]D-Fender |
| 03:19.04 | YoMama | anyone here use gizmo with asterisk? |
| 03:20.02 | k-man | this is my first attempt at an ivr http://pastebin.ca/1366808 |
| 03:20.25 | k-man | how can i get it to hangup after the person has left a message in one of the mail boxes? |
| 03:20.42 | [TK]D-Fender | k-man: hangup <- |
| 03:21.27 | k-man | [TK]D-Fender: could you have a look at my ivr dialplan... i know i could make it simpler but not sure how |
| 03:21.48 | [TK]D-Fender | k-man: nothing simpler, only more to add |
| 03:22.18 | [TK]D-Fender | k-man: well there is something you could do, but its really not worth it |
| 03:23.20 | k-man | [TK]D-Fender: well, its not quite working - 2 problems with it. |
| 03:23.30 | [TK]D-Fender | k-man: only? :) |
| 03:23.49 | k-man | after the person presses 1, 2 or 3 during prompts, there is a long delay before the beep sounds to record the message |
| 03:24.03 | k-man | [TK]D-Fender: well - two that are imidiately obvious |
| 03:25.09 | [TK]D-Fender | k-man: for your first, I don't know what else you have in thaqt context, and I don't see the full attempt to use it either |
| 03:25.39 | k-man | ok |
| 03:25.42 | *** join/#asterisk tobias (n=tobias@user-0ce2hu8.cable.mindspring.com) |
| 03:25.43 | [TK]D-Fender | k-man: if it takes a long toime to enter the VM app off the exten you dial its because what you are dialing is not yet identified as unigue |
| 03:25.48 | [TK]D-Fender | unique |
| 03:26.42 | k-man | [TK]D-Fender: ok - so how do i make it unique? |
| 03:28.35 | [TK]D-Fender | k-man: that depends what else is in the context |
| 03:29.00 | k-man | [TK]D-Fender: ok - ill put together a paste |
| 03:33.35 | k-man | http://pastebin.ca/1366819 |
| 03:33.53 | k-man | another problem is that it plays the "invalid" response after they press hash to end the recording |
| 03:34.07 | k-man | s/they/I |
| 03:37.04 | [TK]D-Fender | k-man: You should already see why |
| 03:37.54 | k-man | [TK]D-Fender: well - i see why, but i don't see how to get it to hang up after a message |
| 03:38.02 | [TK]D-Fender | k-man: Next your IVR should be in its own context. People using it can access your voicemailmain, etc unsecured |
| 03:38.12 | [TK]D-Fender | k-man: HANGUP <- |
| 03:38.14 | k-man | [TK]D-Fender: oh - ok, thanks |
| 03:42.34 | *** join/#asterisk Badrobot- (i=BadRobot@cpe-76-173-233-75.socal.res.rr.com) |
| 03:47.12 | *** join/#asterisk shyam_k (n=user@unaffiliated/shyam-k/x-8459115) |
| 03:49.41 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 03:51.45 | k-man | how do i make a catchall rule at the end of my ivr so if they hit an invalid option it will catch it? |
| 03:52.08 | lanning | use "I" |
| 03:52.45 | lanning | err... "i" |
| 03:52.50 | *** join/#asterisk Frogzoo (n=Frogzoo@59.167.238.221) |
| 03:53.16 | [TK]D-Fender | k-man: go read chapter 5 of the book a few dozen more times..... |
| 03:54.03 | [TK]D-Fender | k-man: read up on "Asterisk Standard Extensions".... these are important basics |
| 03:54.43 | k-man | [TK]D-Fender: ok, thanks |
| 03:57.47 | k-man | thanks lanning |
| 03:59.04 | *** join/#asterisk tobias (n=tobias@user-0ce2hu8.cable.mindspring.com) |
| 04:03.33 | k-man | so how do i reduce the timeout in the irv so that as soon as the key is pressed, it responds? |
| 04:04.55 | *** join/#asterisk Frogzoo (n=Frogzoo@59.167.238.221) |
| 04:05.43 | *** join/#asterisk Frogzoo (n=Frogzoo@59.167.238.221) |
| 04:06.39 | [TK]D-Fender | k-man: another answer I've already given you. Put it in a context with out other stuff that prevents your single digit stuff from being unique |
| 04:07.32 | k-man | [TK]D-Fender: well - i did do that, ill paste its current state |
| 04:08.11 | lanning | core show function TIMEOUT |
| 04:08.56 | lanning | google "asterisk digit timeout", first hit. |
| 04:09.26 | *** join/#asterisk hope4every1 (n=arun@61.95.199.46) |
| 04:10.37 | hope4every1 | Hi. |
| 04:11.08 | hope4every1 | Has anybody implemented Vicidial? |
| 04:12.50 | [TK]D-Fender | lanning: wrong approach |
| 04:13.28 | k-man | http://pastebin.ca/1366838 |
| 04:13.59 | hope4every1 | Hi Fender could u help me to install a astGUIclient/Vicidial |
| 04:14.05 | k-man | ok, thats my ivr context plux what i get when i dial a number |
| 04:15.00 | *** join/#asterisk aenaus_ (n=hdgfghf@79.107.12.51) |
| 04:15.02 | [TK]D-Fender | k-man: And you really aren't looking at what is happening in that call |
| 04:15.12 | [TK]D-Fender | hope4every1: No. |
| 04:16.36 | raden_work | whats a softswitch? |
| 04:16.47 | hope4every1 | ok.Do u know ny1 who can help me |
| 04:16.55 | k-man | [TK]D-Fender: can you give me a clue as to what im looking for? |
| 04:17.32 | [TK]D-Fender | k-man: context <- |
| 04:18.02 | [TK]D-Fender | raden_work: http://www.google.ca/search?hl=en&q=what+is+a+softswitch&btnG=Google+Search&meta= |
| 04:18.31 | hope4every1 | Fender could u help me too |
| 04:18.42 | *** part/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 04:19.32 | *** join/#asterisk bsaxon (n=bsaxon@68.117.152.206) |
| 04:19.41 | *** part/#asterisk bsaxon (n=bsaxon@68.117.152.206) |
| 04:20.00 | k-man | [TK]D-Fender: can I have another clue please? |
| 04:20.34 | *** join/#asterisk bl4 (n=bl4qkuba@216.83.129.167) |
| 04:21.36 | dandate2 | alright my current did provider will handle g729, guess i better sign up! |
| 04:22.25 | [TK]D-Fender | k-mlook at what context is being EXECUTED |
| 04:22.32 | raden_work | anyone know of good company for dirt cheap sip origination ? |
| 04:22.36 | hope4every1 | have nybody heard of the astGUIclient |
| 04:23.06 | [TK]D-Fender | raden_work: www.ipkall.com |
| 04:24.14 | hope4every1 | astGUIclient/VICIDIAL |
| 04:24.15 | k-man | [TK]D-Fender: so its not executing myivr context, its executing phones? |
| 04:24.31 | [TK]D-Fender | k-man: ... *duh* |
| 04:26.18 | k-man | [TK]D-Fender: sorry - i just don't get it |
| 04:27.00 | [TK]D-Fender | k-man: what's not to get? You are dialing an exten that is NOT the new "separate" IVR context you just showed me. |
| 04:27.11 | *** join/#asterisk SunnyDP (n=scan@bas7-montrealak-1096597605.dsl.bell.ca) |
| 04:27.46 | k-man | [TK]D-Fender: ah - i see |
| 04:29.56 | k-man | [TK]D-Fender: got it now, thanks |
| 04:30.02 | k-man | [TK]D-Fender: sorry it took me so long |
| 04:46.23 | *** join/#asterisk astrobear (n=cloud@unaffiliated/ibuffy) |
| 04:50.52 | bl4 | so I like how in the asterisk manual it doesn't talk about running asterisk as a non root user until page 295.....after you installed asterisk on page 50.... |
| 04:51.36 | [TK]D-Fender | bl4: And at the same time they are both in the table of contents. |
| 04:51.56 | [TK]D-Fender | bl4: AMAZING. Almost like you can pick & choose what to watch out for... |
| 04:51.59 | bl4 | ^_^ |
| 04:52.36 | bl4 | who reads the table of contents anyways |
| 04:53.29 | *** join/#asterisk Brack10 (n=travis@97.90.56.111) |
| 04:53.34 | Brack10 | Hey there |
| 04:53.48 | hope4every1 | Hey somebody need a help CRM-asterisk integration |
| 04:53.57 | Brack10 | anyone have any experience with hosted PBX solutions? I'm looking at 8x8 and they're offering some pretty compelling stuffs |
| 04:54.01 | hope4every1 | i am not showing off |
| 04:54.32 | Brack10 | hope4every1: You got your CRM integrated? |
| 04:54.43 | hope4every1 | yep |
| 04:54.50 | hope4every1 | but its one way |
| 04:55.12 | hope4every1 | when call comes in popup comes in my crm with customer details |
| 04:55.36 | *** join/#asterisk aenaus (n=hdgfghf@79.107.29.80) |
| 04:57.38 | hope4every1 | somebody want to know about the easiest CRM-asterisk integration |
| 04:58.28 | *** join/#asterisk Brack10 (n=travis@97.90.56.111) |
| 04:58.32 | Brack10 | oops pidgin crashed |
| 04:58.55 | Brack10 | so CRM guy, does your asterisk drill down when someone calls in, or does it click to call? |
| 05:00.16 | hope4every1 | when someone calls in my crm gives callers details.Now i am working on click to dial.which part is that u r interested |
| 05:00.48 | Brack10 | I don't even have asterisk |
| 05:01.05 | Brack10 | but caller details is more interesting to me |
| 05:01.56 | *** join/#asterisk shyam_k (n=user@unaffiliated/shyam-k/x-8459115) |
| 05:01.57 | Brack10 | seems like it wouldn't be that hard... incoming caller ID > Customer database, match, open opportunity/customer |
| 05:02.09 | hope4every1 | ok Brack setting up an asterisk server is no big deal.Just download an asteriskNOW iso file,burn it to cd,drop it in cd-rom,follow the instructions |
| 05:02.25 | Brack10 | I don't even have IP phones yet |
| 05:02.42 | hope4every1 | get X-lite softphones |
| 05:02.56 | Brack10 | ehh the company would never go for it |
| 05:03.18 | Brack10 | I drafted up a sweet deal for a hosted PBX with 8x8...saved me 44% on telco costs |
| 05:03.59 | drmessano | lol |
| 05:04.24 | Brack10 | lol? |
| 05:04.47 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 05:04.59 | Brack10 | drmessano: something wrong with packet8? |
| 05:05.05 | Brack10 | i mean 8x8 |
| 05:05.35 | *** join/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 05:05.35 | *** mode/#asterisk [+o Mog] by ChanServ |
| 05:05.43 | drmessano | Thats like saying "I switched over to $39.95 lines from Comcast and saved 50% on telco costs!" Yeah, except you're still paying 50% more than the rest of us |
| 05:06.14 | [TK]D-Fender | drmessano: I save with Geiko! |
| 05:06.29 | hope4every1 | so brack what exactly r u planning to do with asterisk |
| 05:06.39 | Brack10 | drmessano: oh so using Asterisk as my PBX will save me way more is what you're saying |
| 05:08.35 | drmessano | Asterisk, when taken as part of a diet including whole grain properly priced telco may or may not result in a large decrease in costs, increase in efficiency and reliabily, and erections lasting more than 4 hours |
| 05:08.48 | Brack10 | I think in addition to the phones they want 1194/mo for 4 offices...5 1-800 numbers, 10 DIDs, unlimited LD etc |
| 05:08.52 | drmessano | Please see a doctor if you experience dizzyness, giddyness, or a payraise |
| 05:08.58 | Brack10 | is that a ripoff? |
| 05:09.52 | drmessano | Good god |
| 05:09.53 | [TK]D-Fender | Brack10: inconclusive |
| 05:10.00 | [TK]D-Fender | Brack10: but quite likely |
| 05:10.17 | Brack10 | 58 phones |
| 05:10.17 | drmessano | These fuckin "Packet8 phones" |
| 05:10.37 | drmessano | Reminds me of the rebranded Polycoms marked up 100% |
| 05:10.43 | Brack10 | probably 5 concurrent per office, 10 concurrent MAX |
| 05:10.52 | Brack10 | $200 per phone for the middle of the road |
| 05:11.03 | Brack10 | $1,139.38 per month |
| 05:11.50 | Brack10 | drmessano: so that's a total ripoff then? |
| 05:12.02 | *** part/#asterisk hope4every1 (n=arun@61.95.199.46) |
| 05:12.35 | Brack10 | cuz if I can drop that cost down, asterisk would be my friend |
| 05:13.45 | *** join/#asterisk hope4every1 (n=arun@61.95.199.46) |
| 05:14.00 | hope4every1 | hi |
| 05:14.35 | Brack10 | I think I'll master AGI while we make the transition to IP and then when the contract runs out deploy Asterisk |
| 05:15.01 | drmessano | Asterisk doesnt save you money.. Asterisk is a toolkit which is commonly used for building PBX systems that faciliate access to a variety of IP and/or analog based services which allow you to have a much greater choice over the services you use |
| 05:15.35 | [TK]D-Fender | yup |
| 05:15.41 | Brack10 | right |
| 05:15.43 | Brack10 | I understand that |
| 05:15.58 | drmessano | Point being, Asterisk is not the solution.. it's an enabler for the solution, which is freedom of choice |
| 05:16.39 | drmessano | Theres lots of options out there, and most companies play on making things easy via retarded bundles of services and overpriced products |
| 05:16.42 | Brack10 | well that's great |
| 05:17.03 | Brack10 | I'd love to own the PBX and be able to choose who I get my service from |
| 05:17.17 | drmessano | and have freedom over the devices you use |
| 05:17.26 | drmessano | and not just lockin to $400 phones |
| 05:17.36 | drmessano | or any specific manufacturer |
| 05:17.48 | Brack10 | yeah I want the cool Cisco phones |
| 05:17.53 | drmessano | ..... |
| 05:17.55 | Brack10 | you know the ones on 24 and The Office |
| 05:18.07 | drmessano | Ok, I am going to stab you now |
| 05:18.13 | drmessano | Hold still pls |
| 05:18.16 | Brack10 | lol |
| 05:18.24 | drmessano | No, really.. stop moving.. |
| 05:18.41 | Brack10 | what's wrong with that? |
| 05:18.48 | drmessano | There's phones much cooler than Cisco's |
| 05:18.51 | [TK]D-Fender | cisco = overpriced trouble |
| 05:18.54 | drmessano | Cheaper too |
| 05:18.56 | Brack10 | like what? |
| 05:19.07 | Brack10 | and does it make the 24 ring? :P |
| 05:19.48 | drmessano | On second though, 8x8 is great |
| 05:19.58 | drmessano | [TK]D-Fender: Show him the door pls, and bring me my cat |
| 05:20.15 | Brack10 | I'm just kidding |
| 05:20.25 | [TK]D-Fender | ~IWMWB |
| 05:20.25 | jbot | I WANT MY WEEKEND BACK! |
| 05:20.25 | drmessano | Oh.... |
| 05:20.37 | drmessano | shuts down his weather machine |
| 05:20.42 | drmessano | Err sorry |
| 05:20.56 | harry_v | dr, getting rained on? |
| 05:21.05 | drmessano | no |
| 05:21.17 | Brack10 | no but really, what phones do you like, messano? |
| 05:22.47 | drmessano | The linksys and polycom stuff is good... I like the Linksys phones, but they're not for everyone |
| 05:23.24 | drmessano | Actually, the Linksys stuff is being rebranded into low end Cisco stuff now, so I guess it's not too bad |
| 05:23.37 | Brack10 | heh |
| 05:23.55 | Brack10 | well shit maybe I shouldn't make the mistake of locking in a super inflated hosted deal |
| 05:24.11 | drmessano | Um YEAH |
| 05:24.49 | harry_v | good for them not for you :) |
| 05:25.09 | Brack10 | I just figured Asterisk was super hard to configure and I would be an outcast in the community if I didn't know how to use the CLI very well and couldn't program AGI |
| 05:25.32 | harry_v | you do not need to use agi at all |
| 05:25.39 | harry_v | for a basic running system |
| 05:25.49 | Brack10 | I see |
| 05:25.57 | drmessano | No, you'll be an outcast if you act like a n00b and dont follow directions many of us put into simple, small pieces, and/or you are an argumentative dumbass SOB.. |
| 05:26.01 | drmessano | I think that sums it up |
| 05:26.16 | Brack10 | What if I use the web interface exclusively? |
| 05:26.30 | drmessano | If you want to learn, it doesnt matter how little you know, just be willing to learn and do the work and theres many that will help you |
| 05:26.31 | harry_v | the you would ask in another chanell |
| 05:26.43 | Brack10 | I see |
| 05:27.07 | harry_v | its not that hard to set up a basic asterisks system. how many phones are you setting up? |
| 05:27.15 | Brack10 | 53 across 4 offices |
| 05:27.31 | harry_v | ohh well that goes beyong basic then :) |
| 05:27.39 | Brack10 | yeah |
| 05:27.39 | harry_v | beyond that is :) |
| 05:27.52 | Brack10 | VPN connectivity for intercom |
| 05:27.59 | Brack10 | 2 cable, 2 DS1 |
| 05:28.11 | Brack10 | analogue paging lines |
| 05:28.12 | drmessano | Dont need VPN for telco |
| 05:28.21 | Brack10 | oh I see, just put it in the DMZ |
| 05:28.22 | Brack10 | makes sense |
| 05:28.23 | harry_v | making a script to repeate sip and extentions would help to create repeadative profiles |
| 05:28.26 | drmessano | God no |
| 05:28.30 | Brack10 | oh |
| 05:28.31 | drmessano | DMZ is evil |
| 05:28.35 | drmessano | You open proper ports |
| 05:28.40 | hope4every1 | how do i install astGUIclient.somebody tell me plz... |
| 05:28.49 | Brack10 | ok |
| 05:29.32 | Brack10 | I have to make a script to maintain an intersite extension numbering convention? |
| 05:29.43 | [TK]D-Fender | hope4every1: http://astguiclient.sourceforge.net/scratch_install.html |
| 05:30.35 | harry_v | squirl ate a hole in our next door neghboors unit and the owner accuses us of boring a hole in her cieling. she thought we were remodeling and it was the squirl making all the racket :) |
| 05:30.47 | Brack10 | what kind of script? Perl? |
| 05:31.04 | harry_v | Brack10, what ever you want c/c++ perl |
| 05:31.09 | harry_v | actually perl |
| 05:31.19 | harry_v | or make a c/c++ app to do it. |
| 05:31.21 | Brack10 | hmm ok.... |
| 05:31.50 | [TK]D-Fender | Brack10>I have to make a script to maintain an intersite extension numbering convention? <- no |
| 05:32.05 | Brack10 | oh ok I misunderstood then |
| 05:32.31 | hope4every1 | Thanx Fender.But when i run perl install.pl it asks for server ip.Wheather it shud be asterisk server ip or system on which astGUIclient runs...help me |
| 05:32.47 | Brack10 | harry: you mean to replicate the configs? I'm confused |
| 05:33.16 | [TK]D-Fender | hope4every1: what do YOU think? |
| 05:34.33 | Brack10 | harry_v: ^^ |
| 05:35.01 | hope4every1 | i don't know which one.I am new into this kind of stuff.It seems u can help me..Fender |
| 05:35.33 | lanning | it asked for the server IP, so which one is THE server? |
| 05:35.33 | Brack10 | Hope4every1: you're installing the client...and it's asking for the server IP..... |
| 05:35.41 | [TK]D-Fender | hope4every1: PICK ONE and see what happens |
| 05:36.06 | Brack10 | Hope4every1: since a client is not a server, that leaves you only one choice by process of elimination |
| 05:37.46 | hope4every1 | my asterisk server is on192.168.1.244 and im installing astGUIclient on 192.168.1.242.Is that all rit? |
| 05:38.14 | Brack10 | *facepalm* |
| 05:39.20 | lanning | hope4every1: which one is the client? |
| 05:39.38 | *** part/#asterisk CrazyTux (n=brandon@ip68-111-67-4.oc.oc.cox.net) |
| 05:41.28 | hope4every1 | x.x.x.242 is supposed to be the client...All i know is asterisk server is necessary for a astGUIclient.So i doubt if it is the server ip they are asking |
| 05:42.50 | lanning | hold on... you have 2 ip's the installer of the client is asking for the server ip, so which machine is the server, and what IP is it using? |
| 05:45.26 | hope4every1 | thanx guys Fender..Brack...i am going trying out all possibilities.....Thanks again |
| 05:47.20 | lanning | you only have 2 possibilities. and one of them, you have been calling a server. the client install is asking for the server IP. and you can't put together which IP it is asking for? and why are you dropping the asterisk server IP, again? |
| 05:48.40 | *** join/#asterisk bl4 (n=bl4qkuba@dsl5-ore-167.fiber.net) |
| 05:48.47 | hope4every1 | Hi lanning.now they are asking DB server address..ooh |
| 05:49.40 | bl4 | so I just restarted my computer to check some settings and I realized.. I am sitting in the car because I have a studio apartment and I am trying ot let my wife sleep....so I'm out in the car working on my server which I reboot....which has a loud boot up music play.... ...Sigh... |
| 05:50.49 | hope4every1 | what is a DB server address.Is it the ip of server itself...help |
| 05:51.22 | Brack10 | bl4: dude, you're working in the car so your wife can sleep? |
| 05:51.28 | Brack10 | what a selfish whore |
| 05:51.52 | bl4 | Brack10: yup, lol. I restarted the computer in the apartment |
| 05:52.13 | lanning | that one, I don't know. I have not used astGUIclient. The only DB use, I know of is CDR and RealTime. |
| 05:52.24 | lanning | both are optional |
| 05:52.48 | Brack10 | bl4: mplayer /music/loudassgrunge.mp3 |
| 05:52.54 | Brack10 | you should do that |
| 05:53.05 | bl4 | lol |
| 05:53.21 | bl4 | *gets locked out |
| 05:53.34 | lanning | what was the reasoning for the "loud boot up music"? |
| 05:54.04 | bl4 | well, it isn't a good server, but its my server. It is an old powerpc mac mini |
| 05:54.15 | hope4every1 | thanx lanning.I am going to use asterisk server ip wherever they ask for server address |
| 05:54.20 | bl4 | I guess they thought it would be cute to play music on boot |
| 05:54.35 | bl4 | or whatever that stupid noise is... |
| 05:54.50 | Brack10 | it's been around since the original Mac |
| 05:55.05 | Brack10 | that was pretty damned impressive in 1984 |
| 05:55.16 | bl4 | yeah, too bad it hasn't gotten any better |
| 05:55.18 | bl4 | ;-) |
| 05:55.39 | Brack10 | heh |
| 05:56.56 | bl4 | so does anyone here run asterisk on linux on a powerpc? |
| 05:58.04 | mog | i have |
| 05:58.43 | bl4 | did it work alright or was there anything weird you had to deal with? |
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| 06:34.39 | stabler | powerpc? |
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| 06:37.05 | mog | nothing weird without hw |
| 06:37.17 | mog | some of the dadhi drivers have endian issues |
| 06:37.38 | mog | or had |
| 06:37.43 | mog | i havent run on ppc in sometime |
| 06:49.51 | lanning | it's a mini, I doubt you can plug in the hardware... :) |
| 06:51.12 | mog | heh |
| 06:51.15 | mog | then i wouldnt worry |
| 06:51.24 | mog | loves my little ppc macmini |
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| 12:03.57 | *** join/#asterisk gsiener (n=gsiener@206.48.2.97) |
| 12:04.08 | gsiener | Anyone here not happy with Callwithus.com? |
| 12:08.57 | UQlev | gsiener: there are plenty providers beside callwithus |
| 12:09.25 | gsiener | tell me about it! I'm just trying to find a payg iax provider |
| 12:09.49 | gsiener | UQlev: who are you using? |
| 12:09.50 | UQlev | gsiener: for what region? |
| 12:10.12 | gsiener | I'm in The Bahamas, but T1 goes through Miami |
| 12:10.30 | UQlev | try checkbox then |
| 12:10.33 | UQlev | I like them |
| 12:10.54 | gsiener | thanks, I will |
| 12:11.23 | UQlev | http://www.checkbox.cc/secure/userlogin.aspx |
| 12:11.33 | UQlev | they are based in Dominica |
| 12:11.42 | UQlev | good rates |
| 12:12.30 | gsiener | Cool. We've been using Voicepulse which had been great until they shut down their iax channels. |
| 12:12.51 | gsiener | For some reason our overbearing router/firewall has a much better time w/ iax than sip |
| 12:13.07 | UQlev | gsiener: actually for asterisk doesn't matter iax or sip, you may setup any |
| 12:13.59 | gsiener | I know, It's just that we have a Sonicwall appliance. It supposedly does SIP qos but ends up mangling calls |
| 12:14.06 | UQlev | gsiener: and better keep your asterisk on pure punlic IP not behind router/firewall |
| 12:14.16 | gsiener | yeah, it's on a public IP |
| 12:15.58 | mort_gib | SonicWall Horror!!! |
| 12:16.43 | gsiener | mort_gib: it's been a nightmare! |
| 12:17.16 | mort_gib | gsiener: Change to OpenBSD |
| 12:17.45 | UQlev | mort_gib: pf is even worse of VoIP |
| 12:17.54 | gsiener | The main issue is that we have three different connections that we're load balancing |
| 12:17.55 | UQlev | for |
| 12:17.56 | mort_gib | UQlev: No problem in NATting your asterisk |
| 12:18.12 | mort_gib | Uqlev: Yeah????? How so |
| 12:18.36 | UQlev | mort_gib: I am using siproxd and rtpd |
| 12:18.49 | UQlev | NAT did not work for me |
| 12:19.14 | mort_gib | Ok... I haven't had ANY problems |
| 12:19.37 | UQlev | mort_gib: for what protocol? sip or iax? |
| 12:19.59 | gsiener | thanks for your help guys |
| 12:20.15 | mort_gib | IAX |
| 12:20.33 | mort_gib | I tried SIP too though |
| 12:20.58 | mort_gib | Really like pf ! |
| 12:20.59 | UQlev | aha, that's a matter |
| 12:21.16 | UQlev | mort_gib: I like pf as well but not for sip |
| 12:21.23 | mort_gib | how come?? |
| 12:22.04 | mort_gib | No way I'm leaving my Asterisk BOX with a public IP |
| 12:22.22 | UQlev | mort_gib: somwhere I fount that pf nat is symmetric or so, that is worst for voip |
| 12:22.49 | mort_gib | ?? symetric NAT?? |
| 12:23.18 | UQlev | mort_gib: I don't remeber now there were 3-4 different types of NAT |
| 12:24.00 | mort_gib | Yeah, just read up on it |
| 12:24.51 | UQlev | and pf's nat the one which can't work even with stun |
| 12:25.11 | mort_gib | Then use RDR |
| 12:25.25 | UQlev | rdr for each user? |
| 12:26.03 | UQlev | my asterisk is on public IP and all users behind router/firewall |
| 12:26.43 | mort_gib | static-port |
| 12:26.44 | UQlev | I had to use siproxd and rtp-proxy |
| 12:26.55 | UQlev | it works fine |
| 12:27.38 | mort_gib | Okay, I would have kept the Asterisk server on the same LAN as the clients, but hey |
| 12:27.46 | mort_gib | Depends on setup |
| 12:28.41 | UQlev | mort_gib: I want my clients to connet from home or from netcafes as well |
| 12:29.33 | mort_gib | Okay, I use Asterisk the other way around |
| 12:29.38 | UQlev | it is a way better than to have mobile roaming |
| 12:30.11 | mort_gib | So Clients are local to my asterisk server and I use providers outside the network |
| 12:30.24 | mort_gib | Yes, if you have Internet |
| 12:30.44 | UQlev | mort_gib: I have local and remote clients and a few providers |
| 12:30.57 | mort_gib | I'm in Spain, so finding a connection is ften problematic |
| 12:31.07 | mort_gib | s/ften/often |
| 12:31.26 | UQlev | even from a cafe? |
| 12:31.44 | UQlev | I mean paid one |
| 12:32.26 | *** part/#asterisk stream (n=stream@securemac.com) |
| 12:34.17 | mort_gib | Uhm, Andalusia |
| 12:34.42 | mort_gib | Sure in Malaga, Marbella, but further down the road, more difficult |
| 12:42.31 | shyam_k | i didn't yet get when this "s" extension will work.. i mean what i should do on the softphone to get to there? |
| 12:43.08 | shyam_k | i guess its proper use is when you have a hardphone? |
| 12:45.50 | shyam_k | sticking to "when calls enter a context without a specific destination extension" howz that possible? |
| 12:45.55 | shyam_k | s/sticking to// |
| 12:52.46 | *** part/#asterisk gsiener (n=gsiener@206.48.2.97) |
| 12:53.05 | shyam_k | ahh.. its not yet 10:00am at the office?:) |
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| 12:56.47 | shyam_k | andrewy: hi |
| 12:56.54 | shyam_k | anthm: hi |
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| 13:34.52 | NMR_1122 | Hi everyone, quick question. Is there a variable (or another way to get) the number that was dialed by the outside party to get into my system? I have multiple numbers (via SAP) coming into Asterisk, and the dial plan should play a different message based on which number the customer called. |
| 13:35.42 | *** part/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis) |
| 13:40.17 | _ShrikE | NMR_1122: ${EXTEN} |
| 13:41.09 | NMR_1122 | That works for incoming calls from an external source too? |
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| 13:41.55 | *** part/#asterisk doolph (n=doolph@190.141.68.31) |
| 13:42.58 | leifmadsen | define 'external source' |
| 13:43.13 | leifmadsen | any incoming call that matches a pattern match will be present in ${EXTEN} |
| 13:43.33 | NMR_1122 | a Voip Provider to which my asterisk box is registered? |
| 13:43.37 | leifmadsen | of course |
| 13:43.44 | leifmadsen | ${EXTEN} doesn't care where it came from |
| 13:44.06 | NMR_1122 | ok, thanks! |
| 13:44.38 | *** join/#asterisk doolph (n=doolph@190.141.68.31) |
| 13:45.46 | mosty | NMR_1122: it may depend on your voip provider, especially if you're using SIP |
| 13:46.37 | NMR_1122 | I have two virtual number with the voip provider. when I call either one, only my primary number shows up in the asterisk console |
| 13:46.55 | shyam_k | hi can i get a pointer to a better explanation for extension s? |
| 13:47.25 | NMR_1122 | it says executing [1231231234@incomming_calls:1] |
| 13:47.41 | NMR_1122 | is the number there what I would get from the Exten var? |
| 13:47.49 | shyam_k | i mean i get that extension i works for an invalid input, and so on. but didnt get when extension s works.,. |
| 13:49.15 | NMR_1122 | I though i was for invalid input? |
| 13:50.08 | shyam_k | yeah its for invalid input, like that i didnt get for what extension s works.. |
| 13:50.47 | NMR_1122 | Sorry, I meant the "i" extension |
| 13:55.50 | NMR_1122 | shyam: " When |
| 13:55.50 | NMR_1122 | calls enter a context without a specific destination extension (for example, a ringing |
| 13:55.50 | NMR_1122 | FXO line), they are passed to the s extension. (The s stands for âstart,â as this is where |
| 13:55.50 | NMR_1122 | a call will start if no extension information was passed with the call.)" |
| 13:56.55 | shyam_k | yeah i quoted that here earlier.. i didn't get how i can make a call enter a context without extension |
| 13:57.13 | NMR_1122 | I think that happens automatically |
| 13:57.28 | shyam_k | like when i call from a softphone, all i do is to dial the extension..so how can i miss the extension or its like, this situation never comes for a softphone? |
| 13:57.30 | NMR_1122 | when you set an incomming line to go into a sertain context |
| 13:58.06 | NMR_1122 | I'm not an expert on this, but I think that all internal calls will be using a specific extension |
| 13:58.33 | NMR_1122 | so "s" is for incoming only |
| 13:59.05 | NMR_1122 | like in your SIP conf file, when you create an entry for your VOIP provider, you say context=my_context |
| 13:59.13 | shyam_k | the thing is if i get that point correctly i have one more core point learned about the structure of asterisk:) |
| 13:59.31 | NMR_1122 | then you would use the "s" extension in the dial plan under [my_context] |
| 14:00.59 | NMR_1122 | "s" is also the only valid extension in a macro |
| 14:10.57 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 14:14.54 | mosty | NMR_1122: you can use extensions other than s in macros |
| 14:15.12 | mosty | it's just that the macro always starts at s,1 |
| 14:15.45 | NMR_1122 | oh, ok |
| 14:16.19 | NMR_1122 | I was trying to answer shyam's question about where extension "s" is actually used. |
| 14:19.37 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 14:24.23 | *** join/#asterisk jeff_phillips (n=jeff_phi@209-206-132-46.dyn.centurytel.net) |
| 14:24.54 | jeff_phillips | hi |
| 14:25.13 | doolph | hi |
| 14:25.20 | mosty | shyam_k: when calls enter the dialplan, they start in extension s if there is no other extension information available |
| 14:27.01 | bl4 | so..I was doing some math and correct me if I am wrong...but if you are using one GSM line 24/7 for 30 days you'll use 0.2 GB in 30 days. Does that sound right...or am I blowing smoke? |
| 14:27.48 | mosty | bl4: sip or iax? |
| 14:27.53 | bl4 | sip |
| 14:28.42 | mosty | http://kb.digium.com/entry/35/59/ |
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| 14:33.21 | jeff_phillips | i come up with 9.23 GB / month from mosty's link |
| 14:33.28 | bl4 | mosty, thanks |
| 14:33.38 | jeff_phillips | did i do that right? |
| 14:34.11 | jeff_phillips | 29.2 kbps divide 8 to get KB/sec, multiply 60 x 60 x 24 x 30 |
| 14:34.37 | bl4 | I dunno, open office died on me... |
| 14:34.47 | jeff_phillips | to get kilobytes per month / 1024 = mb per month / 1024 = GB / month |
| 14:35.17 | jeff_phillips | hmm this time I got 9.02 GB |
| 14:35.28 | jeff_phillips | i must not be fully awake yet. Usually my math gets the same answers twice in a row |
| 14:37.38 | [TK]D-Fender | I miss being consistently wrong too ;) |
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| 14:38.07 | jeff_phillips | i must have done it with 31 days the first time or something |
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| 14:39.59 | bl4 | yeah, I came up with 9.02 GB too |
| 14:40.47 | jeff_phillips | a far cry from 0.2 gb |
| 14:40.59 | bl4 | um, yeah... |
| 14:41.22 | *** join/#asterisk shyam_k (n=user@unaffiliated/shyam-k/x-8459115) |
| 14:41.34 | bl4 | now is that one direction, or is that bidirectional? |
| 14:42.31 | jeff_phillips | good question |
| 14:43.06 | mosty | those rates are for a single direction |
| 14:43.28 | [TK]D-Fender | x2 whee! |
| 14:43.28 | bl4 | sweet...so if you have a line in use 24/7 you are gonna eat up 18.04 GB a month |
| 14:44.14 | [TK]D-Fender | G.729 is lighter and generally considered better voice quality |
| 14:45.05 | bl4 | don't you need to pay fees for that? |
| 14:45.54 | [TK]D-Fender | 10$ one shot |
| 14:46.08 | bl4 | oh nice...who do you do that kinda thing through? |
| 14:46.19 | [TK]D-Fender | www.digium.com |
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| 14:46.35 | bl4 | * feels stupid |
| 14:48.59 | bl4 | thanks for everyones help |
| 14:51.09 | rashed2020_ | Anyone know of a FXO fateway with a VPN client? |
| 14:54.22 | shyam_k | what ip i should be giving in the softphone, if the phone and asterisk is on the same system? |
| 14:54.39 | *** part/#asterisk NMR_1122 (n=rahl@adsl-068-209-105-089.sip.mia.bellsouth.net) |
| 14:54.44 | shyam_k | like the registrar in the twinkle or ekiga? |
| 14:55.31 | shyam_k | ok ekiga don't have facility to change the port but twinkle can.. so what ip i should give to get it connect to the home server? localhost? |
| 14:56.26 | [TK]D-Fender | shyam_k: welcome to networking 101 : 127.0.0.1 or another IP on the server |
| 14:59.42 | *** join/#asterisk Failrar (n=Failrar@coffee.ipv6.kaufmann.tc) |
| 14:59.45 | shyam_k | [TK]D-Fender: i guessed it right;-) sorry for that silly one.:) |
| 15:05.13 | coppice | why would someone want to take their chances with a fateway? :-\ |
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| 15:13.56 | rashed2020_ | har har |
| 15:14.05 | rashed2020_ | lol actually |
| 15:14.24 | rashed2020_ | But yea... gateway with a vpn client? |
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| 15:27.03 | j_kroon | any ideas what asterisk would do if multiple L() options were to be passed to Dial() ? |
| 15:27.23 | jaytee | vomit wildly no doubt |
| 15:32.30 | j_kroon | ok well, then I better make sure it doesn't pass it multiple L() options :) |
| 15:40.57 | j_kroon | unbelievable ... |
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| 16:07.40 | [TK]D-Fender | j_kroon: "It"? You mean " YOU" :p |
| 16:08.30 | j_kroon | [TK]D-Fender, fair enough :) |
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| 17:34.02 | snowboarder04 | I don't suppose anyone has the SIP Flash Image for a Cisco 7940/7960 IP Phone v8.11 they could link me to? |
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| 17:47.19 | Qwell | snowboarder04: cisco.com |
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| 17:49.16 | snowboarder04 | Qwell: yeah, no Technical Support Services Agreement though :( |
| 17:49.23 | *** part/#asterisk c0rnoTa (n=c0rnoTa@80.251.113.56) |
| 17:49.50 | Qwell | Then you'd be violating copyright law. |
| 17:49.59 | Qwell | and nobody here is going to help you |
| 17:50.33 | coppice | This why you shouldn't buy Cisco phones |
| 17:50.57 | coppice | they really should have a model 419 |
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| 18:01.36 | *** join/#asterisk leo54 (n=chatzill@187-25-93-181.3g.claro.net.br) |
| 18:04.25 | leo54 | can someone tell me how do asterisk generate the Call-ID that is used in the recorded file name??? |
| 18:04.32 | leo54 | please |
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| 18:05.13 | Dovid | which file name ? |
| 18:06.52 | leo54 | the file name of the recorded calls in asterisk is formed by <queue>-<date>-<time>-<callID> |
| 18:07.20 | leo54 | example: q1002-20090102-080119-1230890472.36640.gsm |
| 18:07.57 | leo54 | in this case call id is 1230890472.36640. How does this number is generated? |
| 18:08.40 | Dovid | i think that is the call id |
| 18:08.45 | Dovid | most likely epoch |
| 18:08.55 | Dovid | with micro seconds |
| 18:09.33 | leo54 | is this number stored in asterisk database? |
| 18:10.07 | Dovid | no. why would it be ? |
| 18:10.13 | [TK]D-Fender | leo54: What "asterisk database"> |
| 18:10.15 | Dovid | ~epoch |
| 18:10.16 | jbot | 1 Jan 1970 - The expiration date on side of the carton of Unix |
| 18:11.27 | *** join/#asterisk usr13 (n=terry@63.149.173.1) |
| 18:12.32 | leo54 | i want to link a row of a call from db with the recorded file. how do i do that? |
| 18:13.01 | [TK]D-Fender | leo54: WHAT DB? What do you have in it? |
| 18:13.02 | Dovid | in the asterisk cdrs ? |
| 18:13.05 | Dovid | or ur own table ? |
| 18:13.09 | leo54 | cdr |
| 18:13.15 | [TK]D-Fender | Decibel? :) |
| 18:16.20 | leo54 | i want to do like queuemetrics do: display the call informations and a link with the recorded call file name to hear it, but i dont see how to link call info and call record |
| 18:17.01 | Dovid | u can create ur own call logs ;) |
| 18:19.34 | leo54 | but is there a way to get Asterisk Call-ID of a call from asterisk logs? Asterisk uses this call-id to make the name of the call record |
| 18:20.28 | Dovid | from what i remember u can name the file what ever u want |
| 18:20.52 | Dovid | i always used mixmonitor. i put in there what ever i wanted |
| 18:25.24 | [TK]D-Fender | leo54: what do you see in the CDR? |
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| 18:26.36 | *** join/#asterisk HouseMD (n=nandateb@unaffiliated/geek) |
| 18:26.52 | leo54 | where does the asterisk store a call-id? Dovid u said that its epoch with microseconds, can u explain me this? |
| 18:27.22 | *** join/#asterisk bminish (n=bminish@rb.brenbox.home.minish.org) |
| 18:30.35 | leo54 | [TK]D-Fender: in cdr i see the rows: calldate, caller id, src, dst, ..., uniqueID, ... |
| 18:30.56 | [TK]D-Fender | leo54: look at the DATA |
| 18:31.34 | shyam_k | trying with scribblej's asterisk-sphinx pluggin.. neone tried it successfully? |
| 18:32.38 | leo54 | [TK]D-Fender: the row contents? |
| 18:32.52 | leo54 | [TK]D-Fender: i'm seeing them |
| 18:35.46 | *** join/#asterisk mort___ (n=mort@mortier.free-online.co.uk) |
| 18:37.54 | leo54 | {call info's, like date, caller id,...} <= link / associate / find => {call record file} |
| 18:53.22 | *** join/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis) |
| 18:57.31 | *** join/#asterisk RypPn (n=Sally@rosscom.demon.co.uk) |
| 19:05.43 | leo54 | in other words: i want to find the call record file from a registry of cdr table. how to calculate the asterisk Call-ID once the UNIQUEID collumn in cdr is empty? |
| 19:06.46 | [TK]D-Fender | leo54: If its empty then you didn't set up CDR properly to record it |
| 19:29.19 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
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| 19:51.24 | *** part/#asterisk leo54 (n=chatzill@187-25-93-181.3g.claro.net.br) |
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| 20:10.37 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-34-194.w86-215.abo.wanadoo.fr) |
| 20:12.30 | *** join/#asterisk jeff_phillips (n=jeff_phi@209-206-132-37.dyn.centurytel.net) |
| 20:13.21 | jeff_phillips | hi |
| 20:22.09 | *** join/#asterisk implicit (n=bayan@unaffiliated/implicit) |
| 20:31.36 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 20:33.15 | *** join/#asterisk sinein (n=kvirc@c-67-175-114-74.hsd1.il.comcast.net) |
| 20:34.38 | sinein | Hi all, I have some questions related to SS7. Is it always necessary to go through STP's to set up a call? Also, Once the call is set up does a direct SSP to SSP connection take place? Thanks. |
| 20:35.24 | *** join/#asterisk ta^3 (n=tacvbo@201.124.38.103) |
| 20:40.41 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 20:42.33 | ta^3 | I've just installed dahdi-trunk and Asterisk 1.6.0.6 and I'm not able to outbound call using Xorcom's FXO ports until they receive an inbound call. Anyone have a clue regard what's going on? |
| 20:43.20 | tzafrir_laptop | ta^3, I think this was resolved in the latest rc. Let me check |
| 20:43.40 | tzafrir_laptop | A general issue with DAHDI FXOs |
| 20:46.20 | ta^3 | tzafrir_laptop: ok, you tell me. |
| 20:47.02 | tzafrir_laptop | looks like the issue is resolved in 1.6.0.7-rc2 |
| 20:48.02 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 20:49.01 | tzafrir_laptop | ta^3, the fix is http://svn.digium.com/svn/asterisk/branches/1.6.0@160326 |
| 20:49.33 | tzafrir_laptop | that is svn diff -c 160326 http://svn.digium.com/svn/asterisk/branches/1.6.0 |
| 20:51.41 | sinein | no one here knows how SS7 works? |
| 20:55.47 | ta^3 | tzafrir_laptop: very nice! :) i will apply the generated patch to 1.6.0.6. |
| 20:59.50 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 21:02.29 | ta^3 | meh, i see; just need to uncomment the define DAHDI_CHECK_HOOKSTATE. |
| 21:02.56 | drmessano | sinein: At the current, doesnt sound like it |
| 21:06.20 | *** join/#asterisk killown (n=nandateb@unaffiliated/killown) |
| 21:06.55 | *** join/#asterisk HouseMD (n=nandateb@unaffiliated/geek) |
| 21:07.00 | RypPn | tzafrir_laptop I had to add the patch to 1.6.0.7-rc2 for my sangoma to work :( |
| 21:07.28 | tzafrir_laptop | RypPn, what patch? |
| 21:07.49 | RypPn | tzafrir_laptop the one in 14577, http://bugs.digium.com/view.php?id=14577 |
| 21:08.19 | RypPn | dahdi-fxsks-hookstate.patch , as my outbound was getting cahn_unavail again |
| 21:09.21 | tzafrir_laptop | hmmm... my mistake, then |
| 21:09.57 | ajohnson | Anyone know of a way to use the manager interface to put a call on hold? |
| 21:10.22 | ajohnson | Other than redirecting a call to a MusicOnHold command in the dialplan? |
| 21:10.33 | RypPn | I tried the latest 1.6.1 and 1.6.2 and they are the same, unfortunately the patch just makes the calls fail slently, rather than chan_unavail |
| 21:11.11 | tzafrir_laptop | ta^3, right, this is not fixed yet, see http://bugs.digium.com/view.php?id=14577 |
| 21:18.03 | ta^3 | tzafrir_laptop: looks like solved for me. |
| 21:32.58 | *** join/#asterisk blackest_mamba (n=blackest@71.239.160.143) |
| 21:36.09 | *** join/#asterisk mkillebrew (n=fugi@ultra.bl.org) |
| 21:37.06 | mkillebrew | I have sip peer "foo" which has "callerid foo <1015551234>", how do I get asterisk to set its CPN as that number when it dials out? |
| 21:39.37 | mkillebrew | SetCallerID works fine when I do it manually, but I want it to pull from the definitions in sip.conf |
| 21:40.14 | [TK]D-Fender | mkillebrew: It does. Show us your actual config and the CLI output w/ SIP debu for the failed attempt |
| 21:41.04 | *** join/#asterisk mheld (n=mheld@gateway.tippingpointlabs.com) |
| 21:41.07 | mheld | hey y'all |
| 21:41.13 | mkillebrew | it did when I used voicepulse's IAX gateway, it's ceased to since I started using their SIP gateway |
| 21:41.48 | [TK]D-Fender | mkillebrew: put "sendrpid=yes" in your outbound peer entry |
| 21:42.26 | mheld | I'm having issues getting IAX to dial into my phone network |
| 21:42.45 | mheld | I've got two pots lines |
| 21:42.52 | mheld | and IAX with teliax |
| 21:43.04 | mheld | I can get IAX out when both pots lines are in use |
| 21:43.14 | mheld | but I can't seem to get IAX in when both pots lines are in use |
| 21:44.14 | [TK]D-Fender | mheld: that makes no sense, and how is "pots" getting into *? |
| 21:44.42 | mkillebrew | [TK]D-Fender: that did it, thanks. |
| 21:44.47 | mheld | pots -> plain old telephone service |
| 21:45.00 | mheld | we have those hard-wired into the asterisk box |
| 21:45.19 | [TK]D-Fender | mheld: with what?: |
| 21:45.32 | mheld | telephone wire? |
| 21:45.40 | [TK]D-Fender | mheld: .... |
| 21:46.03 | mheld | I have a feeling that i'm screwing up with the extensions |
| 21:46.05 | [TK]D-Fender | mheld: You can't plug a &*^ing wire into SOFTWARE. |
| 21:46.10 | mheld | oh |
| 21:46.13 | mheld | some digium card |
| 21:46.16 | mheld | that's fine |
| 21:46.16 | [TK]D-Fender | better |
| 21:46.17 | mheld | that works |
| 21:46.43 | *** part/#asterisk mkillebrew (n=fugi@ultra.bl.org) |
| 21:46.51 | [TK]D-Fender | mheld: So whats this about "IAX not dialing in"? |
| 21:47.15 | mheld | the call (once there were no pots lines left) used to be forwarded to a number, then the credit card we used to buy the plan expired... |
| 21:47.27 | mheld | so we had to re-update everything through them |
| 21:47.30 | mheld | we got the same number |
| 21:47.35 | mheld | but their server address changed |
| 21:47.59 | mheld | the call should be forwarded to ring on all phones on this network |
| 21:48.09 | mheld | but it seems like it's being hung up |
| 21:48.12 | mheld | automatically |
| 21:48.34 | mheld | when the plan expired, and we had no IAX service, we were getting a "this phone number has been disconnected" error |
| 21:48.47 | mheld | so, that's what makes me think I'm doing something wrong with the extensions |
| 21:50.25 | [TK]D-Fender | mheld: What do you see in IAX debug when a call should come in? what does the IAX2 registry say about your registration status with your provider? |
| 21:50.47 | *** part/#asterisk astrobear (n=cloud@unaffiliated/ibuffy) |
| 21:51.53 | mheld | NOTICE[25538]: chan_iax2.c:6977 socket_read: Rejected connect attempt from 8.14.120.23, who was trying to reach '6175000661@' |
| 21:52.46 | [TK]D-Fender | mheld: Could be a dialplan issue |
| 21:53.23 | [TK]D-Fender | mhenenable full IAX2 debug to confirm that they are authing as the peer they should and confirm the context it points to and its contents |
| 21:53.50 | *** join/#asterisk UQlev (n=kvirc@91.184.221.31) |
| 21:57.28 | mheld | [TK]D-Fender: http://pastebin.com/d47c0050e |
| 21:57.31 | mheld | is what i'm getting |
| 21:59.33 | [TK]D-Fender | mheld: As I said, go look at your peer & dialplan |
| 22:00.36 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 22:00.50 | *** join/#asterisk p1mrx (n=paul@mediabox.apt.pmarks.net) |
| 22:01.00 | mheld | extensions.conf? |
| 22:01.07 | sinein | does anyone here know anything about SS7? |
| 22:01.55 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 22:02.35 | p1mrx | so, I've got an "exten => s/5551212, ..." for calls from one source, and an "exten => s, ..." for all other calls. Is it possible to Goto() the all-callers rule from the caller-id-matching rule? they're both extension "s". |
| 22:03.05 | p1mrx | Goto(s, 1) doesn't appear to work |
| 22:14.40 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 22:15.03 | Assid | err question regarding DID's |
| 22:15.46 | Assid | if you port a number from vendor A to vendor B, does vendor A have to accept the transfer? |
| 22:16.24 | p1mrx | (to answer my own question, it looks like "s/number" and "s" are really part of the same extension, so there shouldn't be two priority 1's) |
| 22:22.55 | [TK]D-Fender | p1mrx: separate your setup into different extens and/or contexts |
| 22:23.13 | [TK]D-Fender | Assid: Yes, A can make your life liveing hell. |
| 22:27.17 | Assid | [TK]D-Fender: crap.. my life just got a whol lot more complicated |
| 22:35.49 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:39.51 | Assid | [TK]D-Fender: ever dealt with j2 communications? |
| 22:43.29 | [TK]D-Fender | Assid: nope |
| 22:43.39 | *** join/#asterisk Flashtek (n=neil@flashtek-uk.com) |
| 22:43.53 | Flashtek | evening all |
| 22:46.14 | *** join/#asterisk moy (n=moy@CPE001cdfec4cee-CM00080dab8485.cpe.net.cable.rogers.com) |
| 22:46.31 | Flashtek | Does anyone here know, if configured correctly, if there is any reason for a TDM400P to NOT work with a UK phone lines CID service ? |
| 22:47.02 | Dovid | Flashtek: I remember seeing something about that a while back. have a look on google |
| 22:47.25 | mheld | [TK]D-Fender: turns out that when they switched servers... they started looking for a different device header |
| 22:47.27 | mheld | fixed it |
| 22:47.29 | mheld | thanks! |
| 22:47.31 | *** part/#asterisk mheld (n=mheld@gateway.tippingpointlabs.com) |
| 22:48.56 | RypPn | Flashtek It would also assume you have clid enabled with BT |
| 22:49.06 | RypPn | Its not by default |
| 22:50.36 | *** join/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 22:50.36 | *** mode/#asterisk [+o Mog] by ChanServ |
| 22:53.31 | Assid | is there a site to show who owns the number |
| 22:57.23 | Flashtek | RypPn: I have asked for CLID to be turned on.. It's with Virgin though.. |
| 22:59.53 | RypPn | Flashtek do you get clid on an inbound call on a standard display phone on the socket? |
| 23:00.33 | Flashtek | I don't actually have a phone that supports it to test it with.. this is part of the problem.. |
| 23:01.09 | Flashtek | the kids dumped the last one that did down the loo.. |
| 23:01.17 | RypPn | buy a dect? you can always whack it into an ata later to get some use out of it |
| 23:01.25 | ta^3 | tzafrir_laptop, RypPn: thank you guys. |
| 23:01.56 | Flashtek | the dect I have doesn't support CID.. Caroline failed to discuss the purchase with me.. |
| 23:03.35 | RypPn | what was her purchase criteria? under 3 quid or she's not playing? |
| 23:03.48 | Flashtek | she was paying.. |
| 23:03.52 | Flashtek | 'nuff said.. |
| 23:05.11 | jeff_phillips | Anyone familer with A2billing's "Least cost dialing" feature. I understand least cost routing -- cheapest route I can buy. But LCD or least cost dialing it says is the "cheapest retail rate (selling rate)" -- what advantage does this give me? |
| 23:06.11 | jeff_phillips | why would I want to sell for as cheap as I can? I thought I'd be trying to make money. |
| 23:06.43 | Flashtek | buy for as cheap as you can, sell for as much as you caan.. |
| 23:07.17 | jeff_phillips | ... then why is it two seperate options? |
| 23:07.29 | Flashtek | i dunno.. |
| 23:07.30 | jeff_phillips | don't you always buy the cheapest rate you can and sell the same call at the highest price you can? |
| 23:07.36 | Flashtek | hides back under his rock.. |
| 23:07.47 | jeff_phillips | I don't get it |
| 23:08.26 | jeff_phillips | unless they are talking about like a resale contract where you get a commission, and it's judging it based on your profit margin... |
| 23:17.02 | *** join/#asterisk raden (n=jon@adsl-99-139-235-165.dsl.applwi.sbcglobal.net) |
| 23:17.28 | raden | jeff_phillips, hows it going ? |
| 23:18.50 | jeff_phillips | good, you? |
| 23:20.41 | NMR_1122 | What is the Voip service called where you get multiple numbers and multiple lines on one plan? So when someone calls any of your numbers, it comes through on whichever line is open at the time? |
| 23:21.50 | florz | NMR_1122: That line stuff is the bussiness of your ISP with VoIP, not of your ITSP. |
| 23:21.53 | florz | -s |
| 23:23.47 | NMR_1122 | I'm not sure lines is the right term? |
| 23:23.57 | drmessano | Its not |
| 23:24.08 | florz | NMR_1122: depends on whether you are talking about lines, I guess =:-) |
| 23:24.08 | NMR_1122 | By line, I mean calls at the same time |
| 23:24.09 | drmessano | You get multiple channels |
| 23:24.22 | drmessano | For concurrent calls |
| 23:24.49 | florz | NMR_1122: so, what is "it comes through whichever call is open at the time" supposed to mean?! |
| 23:25.02 | NMR_1122 | so channels is the voip equivalent of a "line"in the traditional system? |
| 23:25.36 | florz | NMR_1122: no, there is no equivalent, except for the actual IP "line", in case there is any and you aren't connected to the internet by radio |
| 23:26.06 | drmessano | Channels are channels.. the only equivalent in telco would be channels on a T1.. |
| 23:26.07 | florz | NMR_1122: there is no point to any equivalent, as the internet is a packet switching network, as opposed to telephone networks |
| 23:26.09 | drmessano | But lines, no |
| 23:27.48 | NMR_1122 | But the number of channels you pay for is the number of people that can make calls at once, just like having 3 traditional phone lines at your business means three employees can make calls simultaneously? |
| 23:27.59 | jeff_phillips | yes |
| 23:28.02 | drmessano | Pretty much |
| 23:28.42 | jeff_phillips | are there any DID providers where I can receive inbound calls via a SIP trunk, and also receive inbound SMS messages in some means over IP ? |
| 23:28.42 | florz | NMR_1122: no - the number of concurrent calls your plan allows for is the number of concurrent calls you can make |
| 23:28.56 | drmessano | Which is what he said |
| 23:28.56 | florz | NMR_1122: or receive, or whatever you are buying |
| 23:29.26 | NMR_1122 | Ok, I think that channels is what I meant then. |
| 23:29.55 | florz | NMR_1122: there isn't any technical entity that allows one to "make one call at a time", and channel is a pretty asterisk-centric term in this regard |
| 23:30.17 | *** part/#asterisk Flashtek (n=neil@flashtek-uk.com) |
| 23:30.19 | florz | NMR_1122: it's just a matter of your contract, not a technical one at all |
| 23:30.20 | drmessano | Thats not true |
| 23:30.38 | florz | drmessano: which part? =:-) |
| 23:30.41 | drmessano | You havent seen an ITSP offer a specific number of channels |
| 23:30.43 | drmessano | Come on now |
| 23:30.53 | *** join/#asterisk werdan7 (i=werdan7@freenode/staff/wikimedia.werdan7) |
| 23:30.57 | drmessano | Just seems to me like youre being difficult to prove some point here, and hes not that far off |
| 23:31.42 | NMR_1122 | Basically, I'm trying to figure out which provider to use (or what the service would be called), that will allow a customer to call our 800 number, and another customer to call that same 800 number at the same time, and have both go through to the asterisk box... |
| 23:31.56 | drmessano | When I purchase a "DID with 2 channels" from my ITSP, I can rx 2 calls at one time.. When they allow me to terminate 2 calls at one time over a peer, thats not far from his analogy of having 2 lines |
| 23:32.10 | florz | yeah, and I have seen BRI being sold as "two lines", so what? |
| 23:32.35 | drmessano | So CHANNELS is not ASTERISK-CENTRIC |
| 23:32.41 | *** join/#asterisk CapriCoRN^80 (n=int@209.8.41.64) |
| 23:32.54 | jeff_phillips | telcos have always used the term "channels" |
| 23:33.03 | florz | well, ok, asterisk- and itsp-marketing-centric, then? |
| 23:33.14 | [TK]D-Fender | florz: no |
| 23:33.19 | florz | jeff_phillips: telcos haven't been selling VoIP, have them? |
| 23:33.23 | florz | *they |
| 23:33.28 | jeff_phillips | No but they do sell "channels" |
| 23:33.43 | [TK]D-Fender | florz: T1 PRI = 23 B **CHANNELS**. Open ass. Remove head. |
| 23:33.49 | jeff_phillips | I had a remote call forwarding # with the local telephone company one time that only came with 2 channels. I purchased additional channels for an additional monthly fee. |
| 23:34.10 | jeff_phillips | They just programmed the switch at their CO to allow additional simultaneous forwarded calls, to how ever many "channels" i wanted to buy |
| 23:34.11 | CapriCoRN^80 | hi |
| 23:34.18 | florz | jeff_phillips: the question was about the right term for concurrent calls through an ITSP - and that's what I was referring to |
| 23:34.18 | [TK]D-Fender | florz: the term is a telco term. "line is vague. |
| 23:34.25 | jeff_phillips | that had nothing to do with VoIP but it was still a telco defined term called "channels" |
| 23:34.30 | CapriCoRN^80 | which is the best open source billing software for asterisk ? |
| 23:34.39 | [TK]D-Fender | ~toywy |
| 23:34.40 | jbot | toywy is probably The one you write yourself. |
| 23:34.43 | [TK]D-Fender | CapriCoRN^80: ^^ |
| 23:34.54 | Kumba_ | CapriCoRN^80: The one you right write yourself |
| 23:35.04 | Kumba_ | doh |
| 23:35.10 | [TK]D-Fender | Kumba_: pwned |
| 23:35.15 | jeff_phillips | Well the term "channels" simply is a generic term meaning the number of simultaneous calls allowed in a call path |
| 23:35.17 | Kumba_ | hands head in shame |
| 23:35.27 | CapriCoRN^80 | did i ask some thing wrong ? |
| 23:35.44 | Kumba_ | Unless you reallllllllllllly like the way someone elses billing software works, you pretty much left writing your own... |
| 23:35.51 | Kumba_ | There's A2B and FreeSide |
| 23:35.57 | Kumba_ | and some others... |
| 23:36.02 | NMR_1122 | aparently I'm the one who asked the wrong question.... |
| 23:36.32 | CapriCoRN^80 | kumba_: well i am just asking about the available good billing software |
| 23:37.09 | Kumba_ | A2B and Freeside |
| 23:37.25 | florz | jeff_phillips: yeah, right, it's a telco term, and makes perfect sense on TDM and FDM for logical connections with fixed bandwidth allocations ... not so much for describing a limit on the number of session states you are allowed to instantiate on some VoIP media gateway |
| 23:37.40 | CapriCoRN^80 | <[TK]D-Fender> CapriCoRN^80: ^^ ??? |
| 23:37.58 | [TK]D-Fender | CapriCoRN^80: LOOK UP |
| 23:38.04 | [TK]D-Fender | CapriCoRN^80: Get a clue. Seriously |
| 23:38.14 | [TK]D-Fender | [19:34]<jbot>toywy is probably The one you write yourself. |
| 23:38.34 | CapriCoRN^80 | i am not good programmer |
| 23:38.36 | florz | jeff_phillips: just as little as "line" makes sense for describing b channels on a BRI, as telcos over here commonly do |
| 23:38.50 | Kumba_ | A2B and FreeSide are decent billing, as long as they do what you need... |
| 23:38.56 | [TK]D-Fender | CapriCoRN^80: I know all too well... it took you MONTHS to simply install * |
| 23:39.07 | Kumba_ | A lot of people end up writing there own though |
| 23:39.08 | jeff_phillips | florz: But it's still relevent in a VoIP world that still makes use of connections to the PSTN because the telcos usually are going to count how many calls they'll let you send to/from them |
| 23:39.10 | [TK]D-Fender | CapriCoRN^80: I pity your prospective clients |
| 23:39.30 | CapriCoRN^80 | [TK]D-Fender: well i worked hard on it .. doesnot matter |
| 23:39.36 | [TK]D-Fender | the words "uniquely unqualified" come to mind... |
| 23:39.53 | florz | [TK]D-Fender: you weren't trying to tell me something, were you? |
| 23:39.55 | CapriCoRN^80 | [TK]D-Fender: if you are too good .. you should thanks to God but not try to insult others |
| 23:40.50 | [TK]D-Fender | CapriCoRN^80: Too good? Not possible and I am among many more knowledgeable than myself here. |
| 23:41.29 | CapriCoRN^80 | [TK]D-Fender: well your words show ur attitude |
| 23:41.31 | florz | jeff_phillips: yeah, sure they will count, but why does it make sense to call such a counter limit a "number of channels", and why then not just say "your VoIP plan includes 3 lines"? |
| 23:41.36 | [TK]D-Fender | CapriCoRN^80: However anyone who requires such massive hand-holding just to install *, and then jsut an much to get 1 stupid softphone talking to it, and has no programming skills to speak of should not be running a business venture off * |
| 23:41.40 | CapriCoRN^80 | if i am not genius i can work hard |
| 23:42.05 | CapriCoRN^80 | [TK]D-Fender: i am not running a business venture |
| 23:42.10 | Kumba_ | Man, someone spiked fender's kool-aid tonight |
| 23:42.16 | CapriCoRN^80 | [TK]D-Fender: i am a student and doing it for my learning |
| 23:42.42 | [TK]D-Fender | CapriCoRN^80: Billing for learning? Interesting. |
| 23:43.07 | florz | jeff_phillips: if anything, it is, maybe, a limit on the number of channels you may use, which is quite a different thing from a "VoIP line" "having n channels" or something |
| 23:43.14 | CapriCoRN^80 | [TK]D-Fender: why not .. what you think is not always right |
| 23:43.17 | [TK]D-Fender | CapriCoRN^80: If you wanted to learn, you'd have downloaded and tried this stuff yourself. You continue to have people hand-hold you through everything. |
| 23:43.31 | CapriCoRN^80 | so stop making your own decesion about others |
| 23:44.16 | NMR_1122 | so.... |
| 23:44.20 | CapriCoRN^80 | [TK]D-Fender: well i am trying and when i stuck i asked people and try to learn from my mistakes but that doesnt mean that i am always looking for you |
| 23:44.25 | [TK]D-Fender | I prefer to think of them as "historically backed and mutually accepted evaluations of repeat encounters over the span of a year" |
| 23:44.56 | *** join/#asterisk kerx (n=kerx@adsl-69-104-17-222.dsl.irvnca.pacbell.net) |
| 23:45.00 | [TK]D-Fender | Plenty of others who got to see the outlay your current progress required |
| 23:45.00 | Kumba_ | eats some popcorn and drinks his beer |
| 23:45.13 | [TK]D-Fender | Kumba_: We have an excellent half-time show... |
| 23:45.18 | CapriCoRN^80 | and talking about * for months .. well you are wrong again .. as i told you i am student .. i spent 2 days on * and then i was busy in studies and did try few weeks after |
| 23:45.22 | Kumba_ | I'm hoping for a wardrobe malfunction |
| 23:45.26 | CapriCoRN^80 | and you counted that as whole |
| 23:45.40 | [TK]D-Fender | Kumba_: We got them "wardrobe-optional" |
| 23:45.42 | [TK]D-Fender | :) |
| 23:45.51 | CapriCoRN^80 | [TK]D-Fender: I was not installing * for whole month |
| 23:45.55 | Kumba_ | Amen... now just tell me it's an all-female review and i'm there |
| 23:46.20 | [TK]D-Fender | CapriCoRN^80: I remember a solid week of your coming in here for it, and then 2 weeks just beating away with stupid routing issues |
| 23:46.35 | [TK]D-Fender | Kumba_: ^5 |
| 23:46.43 | CapriCoRN^80 | [TK]D-Fender: well not regularly |
| 23:46.49 | NMR_1122 | What would a service be called where you can have any of your inbound numbers use the next available line/channel/call? |
| 23:46.57 | CapriCoRN^80 | [TK]D-Fender: you can check the log |
| 23:47.22 | [TK]D-Fender | NMR_1122: Typically "line-hunting" , "busy/no-answer transfer", or "cascading" |
| 23:47.25 | CapriCoRN^80 | [TK]D-Fender: first i installed it on my laptop which got ubuntu and then on centos |
| 23:47.26 | florz | NMR_1122: well, as I said, there is no such thing |
| 23:47.50 | [TK]D-Fender | florz: Most certainly is. |
| 23:47.55 | florz | NMR_1122: a call is what is created when a user/caller requests it, it doesn't exist beforehand |
| 23:48.26 | florz | NMR_1122: and "lines" or "channels" don't exist as such with VoIP |
| 23:48.42 | NMR_1122 | Yes, but you usually pay by the "line" or "channel" |
| 23:48.59 | NMR_1122 | whether technically correct term, or not |
| 23:49.03 | Kumba_ | Lines or Channels only exist in terms of termination/origination and in terms of software concurrency limits... |
| 23:49.03 | CapriCoRN^80 | [TK]D-Fender: |
| 23:49.09 | florz | NMR_1122: yeah, that is marketing-speak for "maximum number of concurrent calls" |
| 23:49.27 | florz | NMR_1122: tecnically, that's plain nonsens |
| 23:49.28 | florz | +h |
| 23:49.29 | [TK]D-Fender | NMR_1122: From a VoIP perspective there is no inherent limit to the # of channels you can have aside from restrictions placed upon you |
| 23:49.31 | florz | +e |
| 23:49.34 | florz | grrr |
| 23:49.42 | *** join/#asterisk bsaxon (n=bsaxon@68.117.152.206) |
| 23:50.10 | NMR_1122 | Ok, let me ask the question in a different way.... |
| 23:51.58 | jeff_phillips | if the call originaties and terminates entirely on VoIP then channels is only a marketing term. If the call is connected to the PSTN then channels may refer to the number of cocurrent calls allowed through the interconnection point equipment / circuit / magic |
| 23:52.46 | florz | jeff_phillips: as we are in marketing-land, channels may just as well refer to maximum number of concurrent voip-only calls ... |
| 23:53.14 | Kumba_ | PSTN magic is what you want |
| 23:53.18 | Kumba_ | it makes all things possible |
| 23:53.32 | NMR_1122 | I have a working asterisk box. I have an 888 number with the local phone company, which i can port. Right now, we can only take one call at a time because the 800 number is attached to a single phone company line. If a customer calls, and we're talking to them, and another customer calls, the second customer goes to voicemail or gets a busy signal. |
| 23:54.12 | NMR_1122 | We'd like for any calls made to the 800 number to get routed to asterisk, at the same time |
| 23:54.34 | NMR_1122 | (and switching to viop in the process) |
| 23:54.46 | jeff_phillips | NMR_1122: Why do you have telco voice mail on your POTS line? If you absolutely must keep the POTS line, why not setup busy-call-forwarding to send the calls to a DID over a SIP trunk when the POTS line is busy |
| 23:55.10 | jeff_phillips | (of course porting the 800# to a voip service provider is probably wise too) |
| 23:55.22 | [TK]D-Fender | NMR_1122: Point/port your 800# to an ITSP and the only limit will be bandwidth and your agrement with them. |
| 23:56.10 | NMR_1122 | That's sort of the plan... we just aren't sure how to "Google" for the kind of company we want. Voip returns to many consumer-residential type results |
| 23:56.19 | [TK]D-Fender | NMR_1122: A phone # is just a stupid #. Limits are based on agreements and technology. Use of an analog line restricts you to 1 call. VoIP does not. |
| 23:56.24 | [TK]D-Fender | ~itsp |
| 23:56.25 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 23:56.28 | [TK]D-Fender | NMR_1122: ^^^^ |
| 23:56.32 | [TK]D-Fender | ~itsplist-us |
| 23:56.33 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 23:56.38 | [TK]D-Fender | NMR_1122: Get chopping |
| 23:56.43 | [TK]D-Fender | shopping* :) |
| 23:57.01 | florz | NMR_1122: well, just look for an ITSP that does allow enough concurrent inbound calls for your needs - either directly for terminting the toll-free number, or as suggested by others, for forwarding your overflow-calls to, just remember that you'll have to pay for the second (forwarding) call, too, then, probably |
| 23:57.07 | jeff_phillips | gotta go, ttyl |
| 23:58.55 | NMR_1122 | so if i use an ITSP with the toll free number, we don't have to use forwarding then, because the ITSP handles that? |
| 23:59.06 | NMR_1122 | bye jeff |
| 23:59.11 | NMR_1122 | thanks |