| 00:01.18 | *** join/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 00:02.32 | Strom_M | b1ch0: the PSTN uses G.711 |
| 00:08.19 | *** join/#asterisk BBHoss (n=hoss@146.229.191.76) |
| 00:09.42 | b1ch0 | i know , so if * recive an internal call (from ip phone hat have enabled GSM) directed to FXO port, * transcode from GSM to G711 ... right ? |
| 00:09.59 | BBHoss | yes |
| 00:10.29 | b1ch0 | and that is why i have to pay licence for every transcoded call if i decide to use G729 |
| 00:10.38 | JT | Strom_M: analogue uses analogue though, so probably SLIN |
| 00:10.55 | JT | b1ch0: your ip phones support sip and iax2? |
| 00:12.17 | JT | asdx: sip works fine behind nat when setup right most of the time |
| 00:12.28 | b1ch0 | yes both protocols |
| 00:12.48 | JT | b1ch0: must be some crappy chinese designed phone ;) |
| 00:14.45 | b1ch0 | yes they are, but chinese are not stupid ... they copy very well and often improve products |
| 00:15.01 | *** join/#asterisk delmar (n=delmar@ip-58-28-158-154.ubs-dsl.xnet.co.nz) |
| 00:15.36 | JT | b1ch0: considering there are no decent IAX2 ip phones... what do they have to copy? |
| 00:16.05 | *** join/#asterisk xsanchito (n=jorgito@150.138.broadband6.iol.cz) |
| 00:16.06 | xsanchito | hi |
| 00:16.10 | xsanchito | one question |
| 00:16.22 | xsanchito | is asterisk able to act as skinny client to cisco call manager ? |
| 00:16.24 | b1ch0 | knows cisco and huawey story ? |
| 00:17.04 | b1ch0 | anyway, thanks for answering my question |
| 00:17.05 | [TK]D-Fender | xsanchito, IIRC, no. Same with MGCP. |
| 00:17.29 | JT | b1ch0: sorry that didn't make sense |
| 00:17.42 | JT | b1ch0: huawei makes second rate DSLAMs, yes, i know thi |
| 00:17.44 | JT | this |
| 00:17.56 | b1ch0 | just need to know what append when a call is transcoded ..... |
| 00:18.04 | JT | ISPs here are binning Huaweis like there's no tomorrow |
| 00:18.12 | xsanchito | [TK]D-Fender, ok, do you know what open source implements skinny as client ? |
| 00:18.21 | JT | and buying proper equipment like alcatel/lucent/ericsson |
| 00:18.23 | [TK]D-Fender | xsanchito, never heard of any... |
| 00:18.51 | xsanchito | [TK]D-Fender, shame I need it ... anyhow thanks a lot |
| 00:18.57 | JT | b1ch0: and how can they copy if there's no decent IAX2 IP phone to copy off? |
| 00:19.36 | b1ch0 | JT: shure that i will need help to enable NAT |
| 00:19.42 | *** part/#asterisk waverly360 (n=waverly@adsl-070-148-122-203.sip.bna.bellsouth.net) |
| 00:19.43 | JT | xsanchito: why can't CCM speak SIP or H.323? |
| 00:19.52 | JT | b1ch0: what? |
| 00:20.49 | xsanchito | JT, you must pay for SIP by default you have only trunks afaik |
| 00:21.01 | b1ch0 | so if you are so certain that * works well with transversal nat .. i will need your help (i didnt make * with transversal nat before) |
| 00:22.13 | JT | xsanchito: well, weigh up the cost of paying for SIP, or writing an SCCP channel driver |
| 00:22.24 | JT | b1ch0: just read this.: |
| 00:22.26 | JT | ~sipnat |
| 00:22.27 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 00:22.30 | JT | the first url |
| 00:28.55 | b1ch0 | JT: seem interesting .. i will try, i hope next week .... it is the same problem i have ... thanks again |
| 00:31.11 | JT | b1ch0: which problem is that? |
| 00:37.32 | *** join/#asterisk infernix (i=nix@unaffiliated/infernix) |
| 00:38.06 | infernix | has anyone tried wifi sip phones with asterisk, or can anyone comment on how well (or not) wifi sip phones work in general? |
| 00:38.26 | BBHoss | not well |
| 00:38.36 | atomicd | wifi sip phones suck. |
| 00:38.47 | Mackes | I have used alot of WiFi Phones with Asterisk |
| 00:38.55 | Mackes | They dont suck |
| 00:38.56 | BBHoss | SIP DECT phones rock! |
| 00:39.07 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-b8b81fba558fc2f1) |
| 00:39.11 | BBHoss | compared to what? |
| 00:39.23 | atomicd | Which WiFi SIP phones have you used that doen't suck? |
| 00:39.56 | atomicd | mackes: make / model ?? |
| 00:40.01 | Mackes | The Hitachi is very good |
| 00:40.01 | BBHoss | all i have used have slow interfaces, shitty range, low talk time (high battery usage), etc |
| 00:40.12 | Mackes | The Zultys WIP2 is great |
| 00:40.26 | Mackes | The Linksys WiFi300 is ok |
| 00:40.45 | Mackes | Its your WiFi network that makes the difference |
| 00:40.51 | Mackes | Not so much the phone |
| 00:40.57 | atomicd | haha... the Linksys WIP300 sucks donkeys... |
| 00:41.03 | Mackes | ok |
| 00:41.08 | BBHoss | wifi is not suited for voice IMHO |
| 00:41.25 | Mackes | Well.. I guess you have it worked out. |
| 00:41.30 | BBHoss | too much packet overhead |
| 00:41.40 | BBHoss | jitter, latency etc |
| 00:41.50 | Mackes | I promise that the Zultys WIP2 is great $300 |
| 00:41.51 | infernix | there's dect phones with ethernet base stations, but thats not wifi |
| 00:41.53 | BBHoss | no to mention it can be insecure |
| 00:42.09 | BBHoss | infernix: yes thats what im talking about |
| 00:42.17 | BBHoss | they are loads better than wifi |
| 00:42.24 | BBHoss | much better battery life too |
| 00:42.37 | atomicd | I've got two WIP300s on eBay right now. (Sold another earlier this week.) That will be the end of my WIP300 experience. |
| 00:42.44 | Mackes | Well that is just a normal Radio Phone with an ATA |
| 00:43.03 | Mackes | Check out the Hitachi |
| 00:43.35 | Mackes | http://www.zultys.com/index.jsp?tab=productdetail&product=wip2&detail=datasheet-wip2&type=phones |
| 00:43.36 | infernix | well if dect phones with a sip base station are cheaper, that'll do |
| 00:43.51 | BBHoss | they arent really cheaper, but they are better |
| 00:44.08 | Mackes | If thats the case, pickup a Grandstream ATA, and a phone from Wallmart |
| 00:44.23 | Mackes | You can get it done for less then $75 |
| 00:44.27 | Mackes | Same Result |
| 00:44.58 | BBHoss | give this a look: http://www.2gac.net/training.ppt |
| 00:45.14 | BBHoss | the training guide for Aastra's SIP-DECT solution |
| 00:45.21 | BBHoss | it will fill you in |
| 00:45.23 | stubert | Can someone tell me what causes asterisk to send a 491? |
| 00:45.38 | atomicd | Anyone ever try the Polycom/Kirk 600v3 with the Kirk DECT handsets? Saw it at Astricon...looks cool. |
| 00:46.42 | BBHoss | i have never used them, because when i was looking at wireless stuff, they were not avaliavle in the us |
| 00:46.56 | infernix | okay, so ideally i'd need 4 dect phones which are each uniquely addressable |
| 00:46.59 | BBHoss | i have heard from guys at digium that they are a PITA to setup |
| 00:47.07 | infernix | are there base stations that can handle 4 or more concurrent SIP connections? |
| 00:47.13 | infernix | or should I just get 4 base stations |
| 00:47.24 | BBHoss | yes SIP-DECT can handle 8 per base station |
| 00:47.31 | BBHoss | from aastra |
| 00:47.39 | infernix | is aastra affordable? |
| 00:47.41 | BBHoss | take a look at that link |
| 00:47.46 | infernix | compared to ATA + normal dect |
| 00:47.46 | BBHoss | what is affordable to you |
| 00:48.04 | infernix | $400 total was what i'm thinking :) |
| 00:48.06 | BBHoss | if you want a kludgy solution like that, go for it |
| 00:48.17 | BBHoss | ~cheap |
| 00:48.18 | jbot | i guess cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 00:48.53 | BBHoss | a single indoor RFP will cost you probably $600, and the phones are about $250 a piece |
| 00:49.44 | BBHoss | is this for a business or for home? |
| 00:50.36 | infernix | startup business |
| 00:50.43 | infernix | so i just cant shell out $1k |
| 00:50.46 | BBHoss | yours or someone elses? |
| 00:51.03 | infernix | complicated, but lets say ours |
| 00:51.06 | BBHoss | ok |
| 00:51.22 | JT | Mackes: NEVER use grandstream.... what the hell, don't recommend grandstrem |
| 00:51.36 | JT | Mackes: and it is NOT the same result |
| 00:51.37 | BBHoss | do the phones HAVE to be function everywhere, or just in the general area around people offices |
| 00:51.41 | JT | DECT > Wifi |
| 00:51.55 | JT | using wifi for mobile voip is a bad idea |
| 00:52.05 | infernix | the office is like 25 by 5. but it contains highly isolated cabins |
| 00:52.12 | infernix | full metal cages |
| 00:52.18 | BBHoss | heh |
| 00:52.23 | BBHoss | forget wireless then |
| 00:52.27 | BBHoss | of any kind |
| 00:52.34 | infernix | yeah you got a point there |
| 00:52.48 | BBHoss | if wireless is needed inside the offices |
| 00:52.57 | BBHoss | get everyone a 480i CT |
| 00:53.32 | infernix | i think i'll try a regular dect in the soundproof cabins first |
| 00:53.47 | infernix | if fail then ethernet. makes stuff that much more simple, too |
| 00:53.52 | BBHoss | just go to best buy and get a dect phone |
| 00:54.03 | BBHoss | see what range you get with it outside the offices |
| 00:54.46 | *** part/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 00:55.08 | BBHoss | and why do you have full metal cages? |
| 00:55.31 | BBHoss | paranoid of NSA? |
| 00:55.55 | infernix | sound |
| 00:56.02 | infernix | well its not full metal, but a hell of a lot |
| 00:56.05 | BBHoss | METAL? |
| 00:56.08 | infernix | sound isolated cabins |
| 00:56.22 | infernix | metal panels with sound insulators, outside is full metal |
| 00:56.32 | infernix | inside is wood, inbetween a huge insulator layer |
| 00:56.38 | BBHoss | yeah i doubt much signal could get through there |
| 00:56.50 | infernix | i should check my cellphone actually :p |
| 00:57.14 | BBHoss | yeah see how much the signal drops |
| 00:57.39 | *** join/#asterisk mordaunt (n=mordaunt@unaffiliated/mordaunt) |
| 00:59.21 | JT | just wire them |
| 00:59.31 | JT | much more features on wired phones |
| 01:04.44 | *** join/#asterisk relic-se (n=root@82.96.61.130) |
| 01:05.57 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 01:06.45 | relic-se | Hi, can someone help me with a config issue I am having, the documentation has me going around in circles. |
| 01:10.18 | Mackes | Grandstream is not awful, just a pain. |
| 01:10.33 | Mackes | WiFi phones work well, if your AP is good |
| 01:10.53 | Mackes | if you would like to roam between AP's then your AP's need to support |
| 01:10.54 | Mackes | it |
| 01:11.05 | Mackes | Linksys makes the best ATA |
| 01:11.12 | JT | grandstream phones are awful |
| 01:11.16 | *** join/#asterisk MrTelephone (n=test@S010600195b946fa3.ls.shawcable.net) |
| 01:11.19 | MrTelephone | qwell |
| 01:11.21 | Mackes | They are cheap |
| 01:11.27 | Mackes | yes |
| 01:11.31 | JT | wifi phones if you are close to an ap, or you have tonnes of aps |
| 01:11.39 | JT | i don't care how much they cost |
| 01:11.41 | JT | they are awful |
| 01:11.44 | Mackes | ok |
| 01:11.53 | MrTelephone | who wants to make 30 bucks |
| 01:11.54 | Mackes | What do you like Mr JT |
| 01:12.08 | JT | and a Polycom IP320 is very similar cost to a Grandstream GXP2000, and a million times better |
| 01:12.50 | Mackes | Yes, It is a great phone. No doubt, and I am not recomending anyone go by a Grandstream phone |
| 01:12.54 | relic-se | is there any reason why the "s" extension would get a match when no other extensions match? |
| 01:12.59 | Mackes | However, they do have a place |
| 01:13.03 | relic-se | would not get a match even |
| 01:13.47 | JT | Mackes: in the bin |
| 01:13.59 | Mackes | Man, ok |
| 01:14.04 | MrTelephone | relic-se, use i for invalid extension |
| 01:14.43 | Mackes | I like Aastra 480i and 57 iPhones we have hunderds, I also love my Snom 370 |
| 01:14.55 | JT | Mackes: the really is no point supporting cheap shit when the market leader makes products that are very similar in price |
| 01:15.11 | Mackes | Cisco 7960's are the best for a Simple rock hard phone |
| 01:15.16 | JT | i haven't seen the 370, but most snoms were unimpressive |
| 01:15.19 | JT | lol cisco |
| 01:15.33 | Mackes | So Polycom or nothing huh? |
| 01:15.45 | JT | aastra is fine |
| 01:16.06 | [hC] | i understand snom underwent significant changes recently |
| 01:16.28 | [hC] | specifically in the physical design |
| 01:16.34 | Mackes | Snom 370do GSM, and true OpenVPN |
| 01:16.58 | relic-se | MrTelephone: that does not seem to help: Extension '800' in context 'incoming' from '' does not exist. Rejecting call on channel 0/1, span 1 |
| 01:17.09 | Mackes | And Cisco are rock, Rock, Rock solid... Set them up once, and never worry again |
| 01:17.33 | Mackes | My polycom reboots when ever it hits the smallest error with SIP |
| 01:18.43 | Mackes | If I call a Linksys ATA FXO, some times my Polycom reboots |
| 01:18.50 | JT | it must have bad firmware |
| 01:19.00 | Mackes | It locks first and then reboots |
| 01:19.13 | JT | cisco are also cunts :) |
| 01:19.18 | JT | and the audio quality isn't as good |
| 01:19.36 | Mackes | Really? You really dont like Cisco? |
| 01:19.46 | JT | most people don't |
| 01:19.57 | JT | they have terrible business policies |
| 01:20.07 | JT | ridiculous licensing |
| 01:20.46 | Mackes | They released SIP 8.1 for Cisco 7960's to the public a few weeks back |
| 01:21.01 | Mackes | I dont love Cisco as a company |
| 01:21.14 | Mackes | <PROTECTED> |
| 01:21.17 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 01:21.31 | Mackes | However, no one has ever been fired for buying Cisco |
| 01:22.13 | Mackes | So, Asterisk PBX, and Cisco SIP Phones.... Its going to be rock solid, and none techincal users love the Cisco phones |
| 01:22.27 | Mackes | because they see then in Sit Coms, 24, and the west wing |
| 01:22.59 | Mackes | I think we have all wanted a decent used Cisco switch if we could get our hands on one, right? |
| 01:23.42 | Mackes | I have 4 Cisco 7960's at home as home phones.. they have never locked... so I am not worried that my wife will call 911 some day and have the phone lock |
| 01:24.37 | Mackes | Same with my business users, Phones just HAVE to work. This is not to say that Polycoms dont. They are great. But, Polycom is a BIG company as well. |
| 01:26.14 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 01:26.59 | JT | Mackes: no, i avoid cisco switches when i can |
| 01:28.30 | delmar | love my Polycom phone. |
| 01:28.35 | delmar | wish I had more |
| 01:28.50 | Nugget | cisco phones are a royal pain in the ass. |
| 01:28.54 | *** join/#asterisk Cooner750 (n=Cooner75@cpe-71-72-211-147.cinci.res.rr.com) |
| 01:28.56 | Cooner750 | hello. |
| 01:29.48 | Cooner750 | just got Asterisk up and running with Openfire in about an hour, appears to be working rather well so far. |
| 01:30.06 | Cooner750 | I have a question though, it's not possible for users to make outbound calls in any way, is it? |
| 01:30.27 | Nugget | No clue. I've never even heard of openfire. |
| 01:30.47 | Cooner750 | Oh, Openfire is the Jabber server by Jive Software, it dosen't allow the user to dial a number |
| 01:30.51 | Mackes | ok.. well. like I said Polycom are good. Not Cool, nor fun.. But they do the job |
| 01:31.19 | Cooner750 | I'm using Asterisk just for VoIP, no hardware involved. Does anyone else use it like this? |
| 01:31.38 | Nugget | sure, many people do. |
| 01:32.02 | Nugget | you can subscribe to a voip termination service that will let you interface with the public phone network if that's what you're wanting to do |
| 01:32.22 | Nugget | http://connect.voicepulse.com/ is one, http://asterlink.com/ is another one. there are about eleventy-million of them. |
| 01:32.32 | Cooner750 | Oh, nah, not too interested in that. The Openfire server is private anyway, used by friends and me as a chat network |
| 01:32.45 | Cooner750 | Interfacing with the public phone network isn't free, is it? |
| 01:32.47 | Nugget | I wasn't sure if that's what you meant when you said "dial a number" |
| 01:33.58 | MrTelephone | can someone help me with some c code? |
| 01:34.06 | JT | Mackes: cisco phones are well and truly overrated |
| 01:34.16 | Cooner750 | I'll probably be using VoIP on Asterisk just to communicate with friends that can use Spark (Jive Software's jabber client, supports the Asterisk-IM plugin) |
| 01:34.45 | Cooner750 | I'm running Trixbox with Java installed (for Openfire) on a 1GHz PIII with 256MB RAM, it's a little... cramped :P |
| 01:34.46 | rob0 | Cooner750: Not free for outbound, but very cheap. And there are free inbound providers. |
| 01:35.27 | Cooner750 | I probably won't ever have a need to interface with the public network, but if I do I know I can now |
| 01:35.31 | rob0 | I pay <US$0.02/minute |
| 01:35.40 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 01:35.52 | JT | Mackes: their screens flicker horribly |
| 01:36.04 | Mackes | JT, You are very sure of yourself. |
| 01:36.10 | Mackes | They do |
| 01:36.12 | Mackes | ? |
| 01:36.24 | JT | of course i am. i have a Cisco 7905 and 7940 in front of me |
| 01:36.38 | Mackes | I have about 20, that are WELL used (we had a call manager) and they are all fine |
| 01:36.50 | JT | and a linksys, a snom, and polycom |
| 01:36.55 | Mackes | Flicker? |
| 01:37.06 | JT | yes i can see a flicker on the lcd |
| 01:37.13 | Mackes | ok then |
| 01:37.20 | rob0 | JT, I bet you rarely talk on those. :) |
| 01:37.24 | JT | must be a bad refresh rate |
| 01:37.29 | Mackes | Did someone from Cisco hurt you once? |
| 01:37.37 | JT | Mackes: what the fuck? |
| 01:37.51 | JT | Mackes: you can't take the fact that cisco aren't all they're cracked up to be? |
| 01:38.10 | Mackes | Well, i just dont agree. |
| 01:38.10 | JT | rob0: long story, we have a few phone systems here, so i do use them a bit |
| 01:38.37 | Mackes | I have one of quite a few phones, and the Ciscos are consistant |
| 01:38.55 | JT | consistantly not as good as some competition |
| 01:39.05 | JT | of course they're nowhere near as bad as grandstream |
| 01:39.09 | Mackes | Aastras 480i nice, but the firmware took a while before it was stable |
| 01:39.56 | JT | and because i'm not a drooling cisco wanker, someone from cisco must've hurt me? :o |
| 01:40.50 | Mackes | Ok then. |
| 01:41.21 | Mackes | I think we have spoken before JT, however, are you in the US? |
| 01:43.02 | MrTelephone | i have to rewrite *__get_header() to take the last Authorization header :( |
| 01:43.57 | JT | Mackes: no |
| 01:44.22 | Mackes | Around were I live and work, we are swimming in Cisco..... Cisco Switches, Routers, Phones, Call Managers, everyone runs Cisco. Every consultant sells and supports Cisco, Everyone runs PIX firewalls, Every shop is mostly Cisco and Microsoft |
| 01:44.36 | Mackes | Now I dont beleave those are the best products |
| 01:44.48 | Mackes | however they are very good |
| 01:44.56 | Mackes | and we are awash in them |
| 01:45.25 | Mackes | I have so many extra Cisco switches and Routers I am tripping over them in my office |
| 01:45.31 | *** join/#asterisk katsuodo (n=musashi@ool-45776f4f.dyn.optonline.net) |
| 01:45.41 | Mackes | they are just the standard in the US |
| 01:45.50 | Mackes | That doesnt make them the best |
| 01:45.55 | Mackes | however they are good |
| 01:45.58 | Mackes | and availible |
| 01:46.33 | *** part/#asterisk stubert (n=stu@techtools.actusa.net) |
| 01:46.39 | Mackes | That said, We purchaced a very extensive Alcatel OmniPBX system, and Alcatel Siwtches |
| 01:46.44 | Mackes | I hate them |
| 01:46.55 | Mackes | so hard to find support and documentation |
| 01:47.10 | JT | lots of places have Cisco |
| 01:47.14 | *** join/#asterisk chendy (n=chendy@121.76.132.123) |
| 01:47.15 | JT | i fail to see your point |
| 01:48.11 | Mackes | Well, why did so many buy Cisco if it sucks? |
| 01:48.25 | katsuodo | how many extensions can be set on a tdm805 card with dsl? |
| 01:48.32 | JT | because popularity is absolutely no indicator or quality |
| 01:48.39 | JT | worst argument ever |
| 01:49.03 | Mackes | all right then |
| 01:49.28 | Mackes | I understand you point of view |
| 01:50.04 | MrTelephone | don't be subdued by the nonsense of popularity doesn't mean quality |
| 01:50.09 | Mackes | What sold you on Polycom? What happened that made you such a dedicated fan? I like them.. but you are in love |
| 01:50.25 | MrTelephone | there was a cute girl at polycom who helped him setup a phone |
| 01:50.31 | JT | i'm not in love, they are just the best SIP phones available |
| 01:50.38 | JT | ask most people here |
| 01:50.41 | MrTelephone | i agree that polycom are the best phones |
| 01:50.43 | JT | it's common knowledge |
| 01:50.54 | katsuodo | I too agree |
| 01:51.09 | MrTelephone | polycom should beef up their style a little bit |
| 01:51.19 | MrTelephone | polycom phones have nontilting stands |
| 01:51.27 | MrTelephone | huge mistake by the design staff |
| 01:51.36 | MrTelephone | why no backlights? |
| 01:51.42 | MrTelephone | whats wrong with these companies |
| 01:51.51 | MrTelephone | i have a backlight on my 1980 cordless |
| 01:52.13 | JT | heh, most offices have lights in them |
| 01:52.20 | JT | most desk phones don't have backlights |
| 01:52.26 | JT | but the IP650s do i believe |
| 01:52.35 | katsuodo | need info to make right decision, how many extensions can be set on tdm805 card? |
| 01:52.40 | MrTelephone | i work with the lights off |
| 01:53.11 | JT | katsuodo: set? |
| 01:53.14 | Mackes | hahahhhh.. ok |
| 01:53.22 | MrTelephone | polycom have tonnes of features.. provisioning features, failover features, within the phone itself |
| 01:53.34 | Mackes | They all do |
| 01:53.37 | katsuodo | JT halo yes |
| 01:53.37 | MrTelephone | polycoms are practically their own sip proxies |
| 01:53.47 | MrTelephone | but I misread or the documentation about vlans is wrong |
| 01:53.49 | JT | katsuodo: i don't know what you mean? |
| 01:54.03 | katsuodo | okay let me explain |
| 01:54.06 | MrTelephone | I set vlans on my phones and it sets the traffic from the PHONE to the vlan, not from the computer behind it |
| 01:54.09 | JT | Mackes: you know cisco ip conference bridges are rebadged polycoms? |
| 01:54.33 | MrTelephone | JT, how can you prove that, as if cisco would admit it? |
| 01:54.34 | JT | Mackes: and cisco ip phones with SIP firmware use intellectual property licensed off Polycom? |
| 01:54.44 | MrTelephone | cisco should stick to making routers and drop everything else |
| 01:54.52 | JT | MrTelephone: it's BLATANTLY obvious for the conference room bridges |
| 01:54.57 | JT | they look the same |
| 01:55.00 | MrTelephone | haha |
| 01:55.02 | JT | but have a cisco sticker |
| 01:55.17 | MrTelephone | polycom uses koss electronics |
| 01:55.19 | MrTelephone | hahah |
| 01:55.45 | MrTelephone | did you guys see that equipment floating around with the mrtelephone sticker on it? its actually linksys |
| 01:55.50 | katsuodo | in office an asterisk server with a tdm805 fxo card only with 8 pots. Only 5 pots will be used. office manager wants 12 extensions from the card |
| 01:56.07 | MrTelephone | haha i cracked myself up |
| 01:56.11 | MrTelephone | noone else is laughing though |
| 01:56.13 | Mackes | great. So, If it has a Cisco Sticker it will sell better then if its branded Polycom? |
| 01:56.16 | Mackes | Interesting |
| 01:56.42 | Mackes | Why would Polycom what to soil there units with a Cisco tag |
| 01:56.43 | JT | Mackes: the sip firmware in cisco ip phones isn't as good as that on polycom ip phones though |
| 01:57.03 | JT | why would polycom say no to millions of dollars in revennue? |
| 01:57.05 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584064.dsl.bell.ca) |
| 01:57.10 | Mackes | I mean if Polycom is better, they shouldnt need to resell to Cisco? |
| 01:57.17 | MrTelephone | people grow trust with brand names.. some people are ALL cisco or ALL polycom.. but i doubt you see a mix of equipment.. if you do you know its some retard running asteirsk |
| 01:57.19 | JT | they're not reselling |
| 01:57.23 | JT | they're OEMing |
| 01:57.44 | Mackes | Just so we are clearm they will sell more "units" if it says Cisco then Polycom? |
| 01:58.22 | katsuodo | JT understand you |
| 01:58.25 | Mackes | Why not leave it named Polycom, and sell Polycom..... Why boost Cisco's name? |
| 01:58.26 | JT | cisco can either make their own conference bridges |
| 01:58.32 | JT | and they won't be as good |
| 01:58.38 | JT | as polycom have decades of experience |
| 01:58.43 | JT | or they can oem from polycom |
| 01:58.50 | Mackes | yes yes.. |
| 01:59.16 | Mackes | Sell it as a Cisco and they will sell more |
| 01:59.23 | Mackes | why would that be? |
| 01:59.39 | rob0 | MrTelephone: We're laughing at you, not with you. ;) |
| 01:59.57 | JT | Mackes: umm, cisco has customers already, ergo, sales |
| 01:59.58 | MrTelephone | :P |
| 02:00.27 | Mackes | ok guys, |
| 02:00.33 | MrTelephone | polcyoms are so diverse.. web access |
| 02:00.41 | MrTelephone | i love the web access |
| 02:00.53 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-0fbba998d28ce251) |
| 02:01.22 | MrTelephone | someonw wanna write some code for me for 50 bucks :-/ |
| 02:01.25 | MrTelephone | 20 minutes? |
| 02:01.51 | JT | MrTelephone: what sort of code? |
| 02:02.20 | *** join/#asterisk hyphenex (n=scott@60-241-65-249.tpgi.com.au) |
| 02:02.24 | MrTelephone | i need to rewrite get_header() in chan_sip so it looks for Authorization header but it looks for doubles and picks the last one if there is double |
| 02:02.25 | hyphenex | What's a good VoIP phone to buy? |
| 02:02.44 | Mackes | There is only one. |
| 02:02.47 | Mackes | Polycom |
| 02:02.51 | Mackes | Thats it |
| 02:03.00 | Mackes | Everything else is subpar |
| 02:03.05 | JT | aastra are pretty good too |
| 02:03.15 | MrTelephone | im not too good at c so i thought i'd just pay someone to do it |
| 02:03.15 | JT | and linksys in certain countries |
| 02:03.24 | hyphenex | I didn't need it to do anything special, just SIP VoIP calls :) |
| 02:03.26 | Mackes | see... how can you say that.... they lock up like crazy! |
| 02:03.35 | JT | Mackes: what? |
| 02:03.42 | Mackes | And do you know who owns Linksys? |
| 02:03.47 | JT | yes |
| 02:03.50 | JT | why does it matter? |
| 02:04.00 | [TK]D-Fender | hyphenex, in AU the best dollar value is likely to be the linksys SPA series |
| 02:04.06 | MrTelephone | jt do you code? |
| 02:04.18 | JT | MrTelephone: only a little bit, i fumble my way around C |
| 02:04.24 | MrTelephone | same here |
| 02:04.25 | hyphenex | Thanks [TK]D-Fender |
| 02:04.47 | MrTelephone | asterisk is really ahrdcore.. so many references to other stuff |
| 02:04.50 | JT | [TK]D-Fender: nah polycom are a reasonable buy in AU too |
| 02:04.54 | JT | for business use |
| 02:05.00 | [TK]D-Fender | hyphenex, Polycom is IMO the best out there, but I know it comes at a steep premium for you on import. |
| 02:05.16 | MrTelephone | polycom phones work better behind nat as well |
| 02:05.18 | Mackes | So if (and when) Cisco buys Polycom, will Polycom still be a good product? |
| 02:05.18 | JT | [TK]D-Fender: the main problem is lack of IP320/330 availability |
| 02:05.20 | [TK]D-Fender | JT : Really? got a good retailer you could refer me to so I can pass that on? |
| 02:05.26 | JT | Mackes: sure |
| 02:05.30 | JT | [TK]D-Fender: Westan |
| 02:05.37 | [TK]D-Fender | JT, links please :) |
| 02:05.43 | hyphenex | Hmm, it might cost a bit to import then |
| 02:05.48 | Mackes | I just bought 3 Voip Polycom Conference room phones. |
| 02:05.55 | MrTelephone | nice |
| 02:05.59 | MrTelephone | i got one of those ip4000 |
| 02:06.01 | [TK]D-Fender | hyphenex, I meant local distribution... |
| 02:06.02 | MrTelephone | big bucks though |
| 02:06.06 | Mackes | They have been a bitch out of the box to set up |
| 02:06.11 | rob0 | grandsuck |
| 02:06.15 | Mackes | and they dont support POE!!! |
| 02:06.22 | MrTelephone | once you get your system working its easy |
| 02:06.28 | MrTelephone | mackes, which ones did you buy |
| 02:06.30 | Mackes | They came with thier own injector? |
| 02:06.47 | MrTelephone | it came with an injector |
| 02:06.55 | Mackes | The 4 prong star... I think they only have one |
| 02:07.11 | Mackes | yeah... but I dont need one, we are all POE |
| 02:07.28 | [TK]D-Fender | MrTelephone, I picked the SoundStation 2W (Wireless analog) + ATA for my company (got it BEFORE * knowing I'd use it with an ATA later). Works great and the wireless rocks |
| 02:07.30 | Mackes | but the phone only supports its own injector |
| 02:07.50 | Mackes | why would they do that? |
| 02:07.55 | JT | [TK]D-Fender: Westan ... www.whitepages.com.au |
| 02:07.57 | MrTelephone | nice, i don't evne know what a 2w looks like |
| 02:08.03 | JT | i'm sure they're googleable too |
| 02:08.12 | JT | [TK]D-Fender: Westan is the NSW distributor |
| 02:08.43 | asdx | can i use asterisk with vonage? |
| 02:08.50 | MrTelephone | ~googleable |
| 02:08.55 | Mackes | with an ATA only |
| 02:09.08 | JT | Mackes: it was probably designed prior to the ratification of 802.3af |
| 02:09.36 | Mackes | Its the only conference room VOIP they offer, and its brand new |
| 02:09.39 | MrTelephone | can you get a good shock from putting a poe cable in your mouth? |
| 02:10.06 | *** part/#asterisk hyphenex (n=scott@60-241-65-249.tpgi.com.au) |
| 02:10.09 | JT | Mackes: i know, what i said still applies. |
| 02:10.21 | JT | MrTelephone: probably not if it complies with 802.3af |
| 02:10.43 | JT | MrTelephone: the device must signal the switch that it is capable of taking power before power is provided |
| 02:10.51 | MrTelephone | next CSI features a victim who died because someone hooked him up to a non-fcc regulated poe switch |
| 02:11.00 | `Sean | lol |
| 02:11.12 | Mackes | http://www.polycom.com/usa/en/products/voice/small_medium_conference_room/soundstation_ip4000.html |
| 02:11.22 | [TK]D-Fender | JT : no website? Can end users buy direct? |
| 02:11.46 | [TK]D-Fender | MrTelephone, 2W looks just like every other boomerang phone they've ever made :) |
| 02:11.51 | [TK]D-Fender | MrTelephone, its just analog :) |
| 02:12.05 | MrTelephone | :P |
| 02:12.15 | MrTelephone | one of the models didn't support sip either |
| 02:12.17 | MrTelephone | u need an ip4000 |
| 02:12.28 | MrTelephone | i think |
| 02:12.38 | Mackes | http://www.polycom.com/usa/en/products/voice/small_medium_conference_room/soundstation_ip4000.html |
| 02:12.44 | *** join/#asterisk asdx (n=diego@adsl-130-56.click.com.py) |
| 02:12.47 | MrTelephone | fender, and you said thats wireless? |
| 02:12.58 | [TK]D-Fender | to tell you the truth a lot of my users just love the IP600 speakerphone. |
| 02:13.02 | [TK]D-Fender | MrTelephone, yup |
| 02:13.02 | JT | [TK]D-Fender: minimum buy is 1 unit |
| 02:13.12 | MrTelephone | the problem i had is i had to have 2 ftp accounts for the downloads because the ip4000 kept crashing with the sip image that i was using on the polycom 501's |
| 02:13.12 | [TK]D-Fender | JT : is that a yes? |
| 02:13.31 | MrTelephone | but sip image 2.2 or something works on both phones good |
| 02:13.35 | JT | [TK]D-Fender: yes |
| 02:13.39 | *** join/#asterisk marcan (i=1337@host214-205.cvd.fit.edu) |
| 02:13.54 | MrTelephone | whats wrong with the 500 phone fender? |
| 02:14.11 | [TK]D-Fender | MrTelephone, Which 500? |
| 02:14.17 | MrTelephone | 500 series |
| 02:14.18 | MrTelephone | :P |
| 02:14.37 | [TK]D-Fender | MrTelephone, Never said there was anything "wrong" with it. |
| 02:14.48 | MrTelephone | mr high class 600 :P |
| 02:15.06 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 02:15.29 | Nivex | got my PAP2T-NA from telephonydepot today. plugged in and working great. :) |
| 02:15.59 | *** join/#asterisk TJNII (n=TJNII@209.234.89.237) |
| 02:16.03 | [TK]D-Fender | MrTelephone, Ah. This was for my company, and since I wanted PoE I calculated the premium of adding the PoE cable to IP 501's and then factored the fact on the 600 had the MicroBrowser at that time, a higher res screen, and more line-keys with lit indicators, and it was an easy choice :) |
| 02:17.00 | MrTelephone | for browsing the web? |
| 02:17.57 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 02:21.16 | [TK]D-Fender | MrTelephone, Nope, I use the MB for queue stats for our call center, custom logo, etc. |
| 02:21.27 | [TK]D-Fender | MrTelephone, corporate directory, and so on. |
| 02:23.36 | katsuodo | Hello TJNII |
| 02:24.31 | katsuodo | I have question? |
| 02:24.42 | BBHoss | shoot |
| 02:25.38 | asdx | how can i configure asterisk with some voip provider? |
| 02:25.38 | Nugget | heh |
| 02:25.58 | asdx | i've just got a voip provider acocunt |
| 02:25.59 | asdx | account* |
| 02:26.02 | katsuodo | <PROTECTED> |
| 02:26.50 | BBHoss | just use IP |
| 02:26.51 | katsuodo | this is odd request, no? |
| 02:27.03 | BBHoss | yeah, actually its more stupidity |
| 02:27.16 | BBHoss | you can't get 12 ports from an 8 port card |
| 02:27.24 | katsuodo | correct |
| 02:27.35 | *** join/#asterisk shmaltz (n=mybox@mail2.dmaven.com) |
| 02:27.47 | katsuodo | they want use dsl |
| 02:27.49 | BBHoss | you could get two FXS modules and then have each of the two ports ring 6 extensions |
| 02:28.09 | shmaltz | OT:silly question here, how do I create an env var that has more than one value like 2 paths? |
| 02:28.10 | BBHoss | ok that doesnt matter if youre only using IP internally |
| 02:28.26 | katsuodo | no effect bandwidth? |
| 02:28.27 | BBHoss | you can still use your POTS |
| 02:28.30 | *** join/#asterisk hohum (n=dcorbe@dhcp64-134-231-231.shs.nyc.wayport.net) |
| 02:28.39 | BBHoss | nope, as long as you are still using POTS |
| 02:28.49 | BBHoss | well, external bandwdith |
| 02:29.05 | Nugget | well you can certainly have 12 ip extensions and only 8 pots lines, you just won't be able to support more than 8 inbound/outbound calls at the same time |
| 02:29.08 | Nugget | that's perfectly sane |
| 02:29.09 | BBHoss | your LAN bandwdith will be affected minimally |
| 02:29.12 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 02:29.47 | katsuodo | Nugget this must be static ip and then divide? |
| 02:29.52 | BBHoss | i think he wants 5 POTS lines with 12 extensions? using the two spare ports as FXS |
| 02:29.53 | Nugget | pardon? |
| 02:30.19 | BBHoss | you're not understanding us |
| 02:30.34 | katsuodo | pardone |
| 02:30.41 | Nugget | desk phones talking to the asterisk server is one thing. |
| 02:30.44 | BBHoss | you dont need to worry about staic ips or anything, the box doesnt even need an internet connection, as long as its connected to your LAN |
| 02:30.54 | Nugget | the asterisk server talking to the public phone network is a second thing. |
| 02:30.58 | katsuodo | correct |
| 02:31.00 | Nugget | they are almost completely unrelated to each other |
| 02:31.35 | Nugget | if your desk phones are "ip phones" then they will not involve the tdm805 at all |
| 02:31.46 | Nugget | they will "talk" to the asterisk server over ethernet |
| 02:31.58 | Nugget | the tdm805 is for connecting to the public phone network |
| 02:33.54 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 02:34.28 | katsuodo | there is mix of ip phones and rca 2543 4 line intercom speaker phone |
| 02:34.57 | asdx | if i want to use asterisk + voip provider, and i want to use iax2, does the voip provider has to support this protocol? |
| 02:35.08 | BBHoss | asdx:yes |
| 02:35.30 | katsuodo | correct internet no fit in this situation it is pots |
| 02:35.32 | asdx | i wonder if vonage does support it |
| 02:35.38 | BBHoss | no |
| 02:35.41 | BBHoss | vonage sucks |
| 02:35.50 | asdx | BBHoss: which one is good? |
| 02:35.55 | BBHoss | you need an ITSP that sells SIP or IAX2 trunks |
| 02:35.58 | asdx | teliax maybe? |
| 02:36.09 | BBHoss | what do you want internal/outbound or both? |
| 02:36.16 | asdx | BBHoss: both |
| 02:36.22 | BBHoss | are you a home user |
| 02:36.28 | asdx | BBHoss: yes |
| 02:36.34 | BBHoss | hmm |
| 02:37.00 | BBHoss | do you want to pay a flat fee per month (unlimited) or pay a cheap per minute rate? |
| 02:37.17 | asdx | BBHoss: flat rate is ok |
| 02:37.28 | asdx | s/rate/fee |
| 02:37.35 | katsuodo | BBHoss there will be 4 rca 2543 4 line intercom speaker phones |
| 02:37.36 | asdx | BBHoss: is not for me, is for a possible customer |
| 02:37.45 | BBHoss | what country? |
| 02:37.50 | asdx | BBHoss: paraguay |
| 02:37.56 | BBHoss | hmm, i was afraid of that |
| 02:38.34 | BBHoss | do they want to do a lot of international calling? |
| 02:38.40 | asdx | BBHoss: yes |
| 02:38.45 | BBHoss | to what countires |
| 02:39.17 | asdx | BBHoss: usa mostly, i think |
| 02:39.33 | BBHoss | do they want a USA telephone number or a paraguay one? |
| 02:39.36 | asdx | BBHoss: i'm telling the customer to drop vonage |
| 02:40.05 | BBHoss | well if youre using asterisk, vonage doesn't work i dont think |
| 02:40.14 | asdx | BBHoss: they have the computer running asterisk in usa, and they want to connect from here (paraguay) and make calls to usa |
| 02:40.24 | BBHoss | OHH i see |
| 02:40.30 | BBHoss | ok |
| 02:40.36 | BBHoss | so essentially you want a US service |
| 02:40.37 | asdx | they want to connect from here to the pbx* |
| 02:40.44 | asdx | yeh |
| 02:40.52 | BBHoss | ok kool, that makes this easier |
| 02:41.15 | asdx | BBHoss: we want IAX because the morons blocked SIP here |
| 02:41.19 | BBHoss | ohh |
| 02:41.24 | BBHoss | so thats a requirement |
| 02:41.29 | BBHoss | check out VoipStreet |
| 02:41.34 | BBHoss | i use them as backup |
| 02:41.39 | asdx | BBHoss: i'm sure i can use openssh with sip/asterisk but i want IAX for now |
| 02:41.39 | BBHoss | they are alright |
| 02:41.44 | asdx | ok |
| 02:41.46 | BBHoss | they do IAX2 |
| 02:41.49 | BBHoss | and SIP |
| 02:41.54 | asdx | nice |
| 02:41.58 | asdx | thx :) |
| 02:42.09 | BBHoss | IAX2 is minority |
| 02:42.23 | BBHoss | others that do it are VoicePulse Connect (I hate them) |
| 02:42.27 | BBHoss | and a few others |
| 02:43.06 | asdx | what about teliax |
| 02:43.31 | BBHoss | never used them, but i hear they can be flakey at times |
| 02:44.01 | BBHoss | im sure for what you want to do they will work |
| 02:44.25 | BBHoss | i set up phone systems for businesses, which CANNOT be flaky or go down EVER |
| 02:44.34 | BBHoss | which i why i use services that provide SLAs |
| 02:45.00 | rob0 | Tunneling UDP (SIP, IAX2) over ssh is possible through a kludge using nc(1). But it's ugly and generally a bad idea. OpenVPN is much better, provides a complete IP connection to the remote host. |
| 02:45.03 | BBHoss | thats odd that SIP is blocked |
| 02:45.09 | cryptnix | anyone here having issues hitting mikrotik.com? |
| 02:45.33 | rob0 | BBHoss: Not odd if the ISP is a telco. :) |
| 02:45.46 | cryptnix | http://www.mikrotik.com ... (for those of you lazy and don't wanna type) :D |
| 02:45.51 | BBHoss | i guess i dont know how well i have it with AT&T :) |
| 02:46.04 | BBHoss | cryptnix: not coming up for me |
| 02:46.26 | cryptnix | thank you |
| 02:47.33 | asdx | i tell my customer to switch the voip provider |
| 02:47.43 | asdx | he is saying "ok i will consider that" |
| 02:47.45 | BBHoss | well you need to test them first |
| 02:47.49 | asdx | "we are looking at it" |
| 02:47.56 | BBHoss | it is loads cheaper than vonage |
| 02:48.11 | BBHoss | but my question is, which server is blocking SIP? |
| 02:48.22 | BBHoss | and how is vonage working if they block sip? |
| 02:48.29 | Mackes | BBHoss, Use Cisco Phones.... You will be very happy with the performace |
| 02:48.33 | asdx | BBHoss: the telco is blocking SIP |
| 02:48.40 | BBHoss | telco where? |
| 02:48.45 | asdx | BBHoss: paraguay |
| 02:48.52 | BBHoss | but the server is in the US |
| 02:48.57 | BBHoss | correct? |
| 02:49.02 | asdx | BBHoss: yes but my isp goes through the telco |
| 02:49.09 | asdx | BBHoss: it's a evil monopoly |
| 02:49.10 | BBHoss | right |
| 02:49.25 | BBHoss | but if the server is in the US, the servers ISP won't block SIP |
| 02:49.48 | BBHoss | it will need to be like this |
| 02:50.05 | asdx | BBHoss: but the SIP packets are passed through the telco and they block it... |
| 02:50.08 | rob0 | A simple peer-to-peer openvpn between the USA colo and Paraguay, you have your SIP. |
| 02:50.15 | rob0 | or IAX2 |
| 02:50.29 | asdx | rob0: yeah |
| 02:50.40 | BBHoss | Paraguay Office/Home<-->IAX Connection<-->USA Server<-->SIP or IAX2 Connection to ITSP<-->ITSP |
| 02:51.05 | asdx | rob0: what about the router in my lan, that's doing NAT, can SIP<->NAT cohexist? |
| 02:51.09 | BBHoss | note, the home/office will need to have an asterisk server installed |
| 02:51.21 | rob0 | ~sipnat |
| 02:51.22 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 02:51.27 | asdx | ok, thx |
| 02:51.27 | BBHoss | for the IAX2 connection to USA |
| 02:51.37 | rob0 | If you're using an openvpn, you would not need NAT. |
| 02:52.24 | BBHoss | OR, you could just change the port that asterisk runs sip on, then tell your phones to use that port |
| 02:52.41 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-106-89-254.hag.east.verizon.net) |
| 02:52.42 | rob0 | oh yeah, since you control both ends. |
| 02:52.44 | BBHoss | because the telco probably isnt doing level7 blocking |
| 02:54.02 | BBHoss | if they were that would take some beefy routers |
| 02:54.16 | BBHoss | filters w/e |
| 02:58.41 | asdx | my customer is asking me "how can i configure asterisk" |
| 02:58.42 | asdx | wtf |
| 02:59.12 | [TK]D-Fender | asdx, Yeah... you should be asking THEM for help ;) |
| 02:59.34 | Qwell | s/help/money/ |
| 02:59.44 | asdx | [TK]D-Fender: he is a ~60 year old guy |
| 02:59.55 | Qwell | s/money/taffy/ |
| 03:00.33 | katsuodo | with age comes much knowledge and wisdom no? |
| 03:00.39 | Qwell | and taffy |
| 03:00.46 | *** part/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 03:01.08 | rob0 | mmmm taffy |
| 03:02.34 | asdx | ~book |
| 03:02.35 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 03:02.41 | asdx | i will give him the link of the book |
| 03:03.05 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 03:05.09 | BBHoss | damn network is taking a dump |
| 03:07.18 | asdx | do you know if teliax supports IAX? |
| 03:07.21 | katsuodo | Book only for asterisk 1.2 new book come soon via tech support digium |
| 03:07.21 | asdx | IAX2* |
| 03:07.27 | BBHoss | yeah they do |
| 03:07.36 | rob0 | telIAX might support IAX :) |
| 03:08.26 | asdx | i seen a asterisk video and they recommended that |
| 03:08.36 | [TK]D-Fender | katsuodo, pay attention. Thats the SECOND EDITION and is for 1.4 <--------- |
| 03:08.57 | [TK]D-Fender | katsuodo, And that is NOT written by Digium. |
| 03:09.13 | asdx | John Todd |
| 03:09.16 | *** join/#asterisk Winkie (i=sd@87-194-8-125.bethere.co.uk) |
| 03:10.58 | *** join/#asterisk nighty^ (n=nighty@sushi.rural-networks.com) |
| 03:12.41 | *** join/#asterisk voipomatic (n=IceChat7@rrcs-70-63-204-32.midsouth.biz.rr.com) |
| 03:13.20 | katsuodo | [TK]D-Fender understood |
| 03:15.36 | voipomatic | HI All, I am running 1.2.18 and am having trouble transferring calls to remote extensions. If we press transfer and dial an extension which is local to our * it will transfer just fine. But if we try to transfer to an extension that dials out over our VOIP carrier the call will not transfer. We can dial the remote extensions from the local phones but cannot transfer a call to them. Any ideas? |
| 03:17.13 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 03:19.09 | De_Mon | asdx are you serious? you want to know if telIAX supports IAX? |
| 03:19.59 | asdx | De_Mon: yeah, im not very familiar yet with all this stuff |
| 03:20.20 | voipomatic | lol De_Mon |
| 03:20.34 | mosty | does the j option to Dial mean "jump" or "don't jump" in asterisk 1.2.13? the wiki page is contradictory on this |
| 03:20.39 | De_Mon | think about it, telIAX... |
| 03:20.45 | De_Mon | tel-IAX |
| 03:20.45 | asdx | ohhhh |
| 03:20.46 | asdx | damn... |
| 03:20.50 | voipomatic | lol |
| 03:20.59 | voipomatic | asdx it happens to the best of us |
| 03:21.03 | De_Mon | asdx you might want to get some rest before you try anything else today |
| 03:21.13 | asdx | De_Mon: indeed |
| 03:21.14 | asdx | :-) |
| 03:21.57 | asdx | i was going to sleep already... |
| 03:22.15 | voipomatic | asdx - TelIAX is a great choice BTW |
| 03:22.28 | asdx | voipomatic: yeah, i recommended that to the customer |
| 03:22.46 | asdx | voipomatic: he said he will decide with that... |
| 03:22.55 | asdx | s/recommended/showed |
| 03:24.00 | voipomatic | yeah, their support is great. (the only thing they will do is try to defer you to the per hour * support, so just be sure, if you call them, to be insistant that the problem lies on their end and they will find the problem) |
| 03:24.52 | voipomatic | they do go out of their way for ya. who you all using now? |
| 03:27.08 | asdx | voipomatic: my customer was using vonage with some client, i don't know what... but i think he didn't had a pbx, but he stopped to use because the telco company blocked SIP. |
| 03:27.17 | asdx | voipomatic: now he called me for an "alternative" |
| 03:27.47 | voipomatic | interesting to hear that sip was being blocked |
| 03:28.09 | asdx | he said he doesn't have money though |
| 03:28.14 | asdx | but i will do this anyway... to learn |
| 03:29.05 | asdx | they want to be ISP |
| 03:29.06 | asdx | lol |
| 03:29.11 | asdx | err |
| 03:29.18 | asdx | voip service provider |
| 03:29.18 | voipomatic | lol, not without any money |
| 03:29.43 | CrazyTux | Anyone here ever used an SPA9000 ? |
| 03:30.09 | asdx | voipomatic: i will ask him for some cash when i have everything running |
| 03:31.09 | *** join/#asterisk JayTee52 (n=jforde05@c-69-137-243-25.hsd1.in.comcast.net) |
| 03:31.36 | asdx | he wants to use asterisk + his voip provider for now, and make his calls |
| 03:32.07 | voipomatic | vonage? |
| 03:32.12 | asdx | teliax |
| 03:32.16 | asdx | he will switch |
| 03:32.48 | asdx | i'll try to convert him to linux |
| 03:32.55 | asdx | he wants to learn asterisk and cli |
| 03:32.55 | asdx | lol |
| 03:33.07 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-227-239.dsl.irvnca.pacbell.net) |
| 03:33.14 | asdx | bash, etc |
| 03:40.21 | asdx | is IAX a standard, or will be one? |
| 03:40.27 | asdx | like SIP |
| 03:40.37 | voipomatic | yes |
| 03:41.12 | asdx | nice |
| 03:41.29 | mosty | it's not quite a standard like sip |
| 03:41.46 | Qwell | oh, that Cap'n Crunch bit on Drawn Together was SO appropriate |
| 03:41.59 | asdx | IETF standard |
| 03:44.16 | Nugget | which is better? normal cap'n cruch or peanut butter? :) |
| 03:44.28 | Nugget | crunchberries are plain disgusting. not even in the running. |
| 03:44.35 | *** join/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 03:44.39 | Mackes | hey |
| 03:45.02 | Qwell | peanut butter, but you've gotta have standard occasionally |
| 03:49.28 | asdx | what is a good IAX hardphone |
| 03:49.55 | Mackes | so, does anyone use a playtone to indication the beginning of a call |
| 03:52.34 | *** join/#asterisk bmg505 (n=leon@196.209.183.81) |
| 03:53.24 | mosty | Mackes, besides the default ringing sound? |
| 03:57.13 | *** join/#asterisk BigCanOfTuna (n=chatzill@dsl-mac-66-18-226-119-cgy.nucleus.com) |
| 04:00.46 | TJNII | asdx: Not too many make iax hardphones |
| 04:00.59 | TJNII | I picked one up on eBay but the DSP is crap |
| 04:01.09 | JayTee52 | I know that X100 cards are not supported anymore and that they are not recommended but can I use 2 of them with a TDM400 card also? |
| 04:01.24 | BigCanOfTuna | I'm trying to pass a variable from a call file to my dialplan, my dialplan has this line: Setvar: <code=911257> and my dialplan has an operation like this: exten => _XX,1,NoOp(${code}) ................the NoOp is being called, but the output suggests that no value was passed...any thoughts? |
| 04:02.11 | mosty | JayTee52, yeah probably. but you should probably get a BRI or fractional PRI for the effort and call quality |
| 04:02.45 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 04:02.51 | mosty | BigCanOfTuna, paste that section of your dialplan on a paste site |
| 04:03.12 | asdx | TJNII: i see |
| 04:03.39 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:03.40 | JayTee52 | mosty, we intend to move all PSTN traffic to Digium's PRI card but for now we just need to add a couple lines and have no slots available so I need to replace 1 X100 card with a TDM400 we just bought. |
| 04:04.45 | mosty | JayTee52, get a sangoma PRI card with hardware echo can instead |
| 04:05.10 | JayTee52 | I'm being squeezed by my boss who is a major [expletive deleted] and lacks the sack to ask his boss for more money until he can WOW them with some fancy dog an pony show crap using Exchange Unified Messaging. |
| 04:05.31 | BigCanOfTuna | mosty: Here you go: http://pastebin.ca/774274 |
| 04:06.17 | JayTee52 | then we can buy some decent hardware and route our existing PRI lines through Asterisk to SIP phones and also to our legacy Nortel PBX. |
| 04:06.49 | mosty | BigCanOfTuna, pastebin.ca looks hosed, doesn't show me anything on that page |
| 04:07.15 | BigCanOfTuna | mosty: You sure, looks good to me in FF. |
| 04:07.36 | BigCanOfTuna | mosty: Looks good in Safari. |
| 04:07.46 | mosty | yeah, it just shows me a blank page in iceweasel/firefox |
| 04:08.00 | *** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60) |
| 04:08.09 | JayTee52 | so from everything I can find in Asterisk TFOT and other sources online am I right in thinking the wctdm Zaptel driver will work with both the TDM400 card and the X100 cards? |
| 04:08.52 | mosty | yes |
| 04:08.58 | mosty | from memory |
| 04:09.26 | JayTee52 | mosty, thanks! |
| 04:09.32 | *** part/#asterisk BigCanOfTuna (n=chatzill@dsl-mac-66-18-226-119-cgy.nucleus.com) |
| 04:13.27 | *** join/#asterisk mitcheloc (n=mitchel@adsl-68-120-227-239.dsl.irvnca.pacbell.net) |
| 04:13.49 | JayTee52 | good night everyone! |
| 04:13.59 | *** join/#asterisk Netgeeks-laptop (n=chris@204.11.231.198.static.etheric.net) |
| 04:14.05 | JayTee52 | and thanks again for your help |
| 04:14.16 | *** part/#asterisk JayTee52 (n=jforde05@c-69-137-243-25.hsd1.in.comcast.net) |
| 04:16.24 | MrTelephone | i never pictured myself as a dog killer but i probably will poison the dog next door if they keep putting it out at 11am |
| 04:17.35 | [TK]D-Fender | MrTelephone, tameshigiri ;) |
| 04:18.29 | MrTelephone | whats that |
| 04:18.38 | De_Mon | is there a dialplan app that will tell me how many people are holding in a queue? |
| 04:19.07 | MrTelephone | exten => 911,1,Dog(Bullet) |
| 04:19.46 | [TK]D-Fender | MrTelephone, jfgi :) |
| 04:21.36 | MrTelephone | hey i have to make a little powerpoint game version of deal or no deal |
| 04:21.36 | MrTelephone | just the money part |
| 04:21.36 | MrTelephone | but i can't remember what the denominations are |
| 04:22.14 | [TK]D-Fender | MrTelephone, ..... jfgi! :) |
| 04:22.48 | De_Mon | http://pastebin.ca/774285 |
| 04:23.10 | De_Mon | [TK]D-Fender there's that crazy idea you helped me with earlier this week |
| 04:23.21 | [TK]D-Fender | De_Mon, Probably did.... |
| 04:24.41 | De_Mon | probably did? |
| 04:26.56 | De_Mon | Here's the thing, at line 36 the caller (customer) is dialing line 50 and after about 5 minuites it stops playing hold music and says Call Failed: Request terminated |
| 04:27.28 | De_Mon | thats the sip phone, I'm tryin it again now... so its almost done |
| 04:28.52 | [TK]D-Fender | De_Mon, pastebin your queue def |
| 04:30.05 | *** join/#asterisk CrazyTux (n=CrazyTux@216-110-94-231.static.twtelecom.net) |
| 04:32.34 | De_Mon | reupdated |
| 04:32.35 | De_Mon | http://pastebin.ca/774295 |
| 04:32.54 | De_Mon | oh, silly pastebin made it sound like it was going to update the previous paste |
| 04:33.02 | [TK]D-Fender | # |
| 04:33.03 | [TK]D-Fender | joinempty = strict |
| 04:33.03 | [TK]D-Fender | # |
| 04:33.03 | [TK]D-Fender | leavewhenempty = strict |
| 04:33.18 | [TK]D-Fender | hrm... if you're not answering I'm thinking you're getting kicked if it caps at 5 min |
| 04:33.43 | [TK]D-Fender | De_Mon, would also be nice to see the call... |
| 04:34.14 | De_Mon | its helluva messy every 10 seconds I get like 10 lines in the cli |
| 04:34.18 | [TK]D-Fender | De_Mon, OMG, parked agents! lol |
| 04:34.38 | [TK]D-Fender | De_Mon, sickening brilliant. This is cute.... |
| 04:35.25 | De_Mon | it was your idea :) |
| 04:35.46 | [TK]D-Fender | De_Mon, umm.... was it? |
| 04:35.51 | De_Mon | something is definately weird, the caller isn't actually getting to the queue |
| 04:36.04 | [TK]D-Fender | De_Mon, that doesn't sound good... |
| 04:36.07 | De_Mon | Yeah I came in with a how do i... and you came up with it |
| 04:36.51 | [TK]D-Fender | De_Mon, if only I could remember all this stuff I come up with... |
| 04:39.25 | De_Mon | you'd be rich! |
| 04:42.08 | De_Mon | http://pastebin.ca/774305 |
| 04:42.54 | De_Mon | Here's the first half, it keeps trying the members and waiting for an agent to accept call for 5min and then hangs up. I've added some extra debugging in hangup to see what I can see |
| 04:43.21 | De_Mon | Okay I just heard the periodic announce, so I guess they are in the queue |
| 04:44.20 | De_Mon | http://pastebin.ca/774307 |
| 04:44.30 | De_Mon | thats show channels output |
| 04:48.04 | De_Mon | http://pastebin.ca/774311 -- And there's the end of the call, dialstatus is cancel? |
| 04:49.14 | De_Mon | oops, forgot my s/elephant/queue/ substitution. |
| 04:49.48 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 04:50.53 | De_Mon | s,8 is canceling the call after 5min. wtf |
| 04:51.31 | De_Mon | exten => s,n,Dial(Local/support@elephant-support2||G(s^pickup)) |
| 04:52.34 | De_Mon | oh thats right, Dial is holding the call in my queue... |
| 04:54.08 | *** part/#asterisk Mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 04:56.01 | [TK]D-Fender | De_Mon, looks like it recurses. |
| 04:56.09 | [TK]D-Fender | De_Mon, I'm getting dizzy |
| 04:56.57 | De_Mon | I have a pile of paper where I planned this out with all sorts of drawings trying to get it straight |
| 04:57.14 | De_Mon | most of them are in the trash |
| 04:58.24 | De_Mon | The agent is called, it goes to the remote centers queue for however long. When someone finally picks up, they press a button to actually get parked and join the queue |
| 04:59.10 | De_Mon | so I'm not really sure why the calling party stays in the queue with 2 static members that arn't really there 99% of the time... |
| 04:59.11 | [TK]D-Fender | De_Mon, Clean it up... I'm all outta dramamine :) |
| 04:59.28 | De_Mon | But the thing is, its the Dial command that is canceling. |
| 04:59.31 | De_Mon | clean it up? that is clean! |
| 04:59.33 | [TK]D-Fender | De_Mon, stays in queue because those are STATIS memebers and can not be kicked. |
| 04:59.42 | [TK]D-Fender | static |
| 05:00.31 | De_Mon | so why is dial canceling the call after 5 min |
| 05:00.44 | De_Mon | I'm pretty sure thats the real issue |
| 05:01.09 | [TK]D-Fender | De_Mon, maybe its that remote queue? |
| 05:01.51 | *** join/#asterisk tecnico (n=tecnico@user-24-214-56-217.knology.net) |
| 05:01.56 | De_Mon | nope, they are still looping thru their little script after the calling party is disconnected. |
| 05:04.02 | De_Mon | I'm testing a normal dial(sip/jon) lets see if it cancels after 5min... |
| 05:04.51 | De_Mon | do you really think there is cleaning up that could be done to this? |
| 05:10.42 | De_Mon | It's been 7min, thats not whats canceling the call. |
| 05:14.19 | De_Mon | now just dialing local/support@queue-support2. It's kinda strange that it's playing music on hold isn't it? |
| 05:15.03 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 05:15.05 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 05:15.45 | De_Mon | Dial doesn't think its done until the call is bridged, I don't think it was designed to call queues :) |
| 05:22.50 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 05:23.13 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
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| 05:29.31 | De_Mon | could the calling phone have sent the CANCEL? |
| 05:31.28 | *** part/#asterisk katsuodo (n=musashi@ool-45776f4f.dyn.optonline.net) |
| 05:32.10 | De_Mon | I doubt it, but I'm testing from a different phone just to make sure |
| 05:40.24 | De_Mon | ahhhh, the softphone still thinks im "calling" |
| 05:40.36 | De_Mon | I bet it timed out... |
| 05:55.29 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
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| 06:01.30 | De_Mon | success! |
| 06:04.01 | [TK]D-Fender | Oh? |
| 06:12.04 | De_Mon | yeah, stupid softphone. Threw an Answer() into the calling side and it didn't CANCEL the call |
| 06:12.17 | De_Mon | it was in ringing state for 5min and canceled the call |
| 06:12.33 | De_Mon | dialing a queue does funny things :) |
| 06:12.39 | *** join/#asterisk Ore4444 (n=ore4444@CBL217-132-245-188.bb.netvision.net.il) |
| 06:12.44 | *** part/#asterisk Ore4444 (n=ore4444@CBL217-132-245-188.bb.netvision.net.il) |
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| 06:45.42 | BBHoss | do any of dialogic's products work with *+ |
| 06:47.25 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 06:52.01 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
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| 07:04.46 | tzafrir_home | BBHoss, I think at least some will work with some proprietary channel driver (requires BE) |
| 07:05.02 | tzafrir_home | Unless you refer to DIVA cards |
| 07:05.16 | tzafrir_home | which should work with the external chan_capi |
| 07:26.59 | *** join/#asterisk Raky-2 (n=John@220.157.75.66) |
| 07:27.12 | Raky-2 | hey guys, i was wondering if it's possible to convert the extensions.conf into mysql statements |
| 07:27.19 | Raky-2 | or at least be able to run asterisk extensions, from mysql? |
| 07:31.19 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
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| 09:03.12 | Chris-NB | hi |
| 09:03.19 | Chris-NB | anyone using q.sig on a pri line? |
| 09:08.08 | *** join/#asterisk MicW (n=michael@dslb-088-074-154-013.pools.arcor-ip.net) |
| 09:08.12 | MicW | hi |
| 09:08.36 | MicW | is anyone here using a siemens gigaset c470? |
| 09:08.48 | MicW | (sip phone) |
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| 09:27.46 | *** join/#asterisk Telemac (n=telemac@213.223.113.74) |
| 09:27.52 | Telemac | Hello |
| 09:28.14 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 09:28.57 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 09:31.09 | Telemac | I'm trying to setup odbc in asterisk to put config in database. I've installed unixODBC, configured odbcinst.init and odbc.ini. This seems ok as I can connect with isql. Then I've added res_odbc.conf and restart asterisk. When I run 'asterisk -r' odbc commands work, so I suppose odbc modules are properly loaded bu 'odbc show' doesn't display dsn configured on res_odbc.conf. Any idea what's wrong ? |
| 09:34.41 | *** join/#asterisk cjk (n=loic@80.92.64.103) |
| 09:35.43 | billybongo | Telemac: what does odbc show tell you? |
| 09:36.14 | cjk | hi, is there a way to store peers (i am using asterisk realtime) in more then one table without using database features like views? |
| 09:36.16 | *** join/#asterisk PepOSX (n=pepOSX@190.72.153.45) |
| 09:37.44 | Telemac | billybongo: nothing |
| 09:38.11 | Telemac | ulysse*CLI> odbc show |
| 09:38.11 | Telemac | ulysse*CLI> |
| 09:42.16 | billybongo | Telemac: but odbc connect seems to work? |
| 09:42.25 | *** join/#asterisk blq (i=me@1.1.1.vg) |
| 09:42.30 | billybongo | run with more verbosity and see if it tells you more |
| 09:42.49 | billybongo | also are your database logs showing connections? |
| 09:45.05 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 09:53.44 | *** join/#asterisk jozu (n=torrent@84.120.184.91.dyn.user.ono.com) |
| 09:54.23 | jozu | hi to all |
| 09:54.47 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-c765bbd5e485b47f) |
| 09:55.05 | Telemac | billybongo: I've already increase verbosity, and it doesn't tell me more |
| 09:55.11 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-c765bbd5e485b47f) |
| 09:56.32 | Telemac | I don't think there could be any connection as firstly there is no dns (while it is configured) |
| 09:58.50 | *** join/#asterisk atari (n=atari@213.144.146.89) |
| 09:58.51 | atari | hi |
| 10:00.11 | billybongo | Telemac: so you mean that there's no way for it to resolve the db server |
| 10:01.20 | Telemac | billybong: I think. It's as if res_odbc.conf is not read whereas odbc module are loaded |
| 10:01.42 | Telemac | preload => res_odbc.so |
| 10:01.42 | Telemac | preload => res_config_odbc.so |
| 10:01.53 | Telemac | I've also tested replacing preload by load |
| 10:04.55 | billybongo | check permissions on the file |
| 10:06.21 | billybongo | also check the messages when restarting asterisk |
| 10:06.33 | Telemac | billybongo: same perms as extensions.conf (working |
| 10:07.04 | Telemac | billybong: No special message when restarting, neither in console nor in logs |
| 10:07.31 | billybongo | what's in it? |
| 10:08.39 | billybongo | you shold have the dsn as defined in your odbc config, username and password |
| 10:09.05 | billybongo | i.e. the bit in [] in odbc.ini |
| 10:09.36 | billybongo | can you connect OK to the dsn outside of asterisk? |
| 10:09.39 | Telemac | billybongo: odbc.ini is ok, as I can connect with isql and expectedn dsn |
| 10:09.44 | billybongo | ok |
| 10:10.07 | billybongo | so in my odbc.ini I have |
| 10:10.07 | billybongo | [asterisk] |
| 10:10.07 | billybongo | ..... |
| 10:11.03 | billybongo | then in my res_odbc.conf I just have |
| 10:11.03 | billybongo | [asterisk] |
| 10:11.03 | billybongo | dsn => asterisk |
| 10:11.03 | billybongo | username => blah |
| 10:11.03 | billybongo | password => secret |
| 10:11.04 | billybongo | pre-connect => yes |
| 10:11.25 | billybongo | Telemac: does yours look like that? |
| 10:12.35 | Telemac | For perms : -rw-r----- 1 root asterisk 100 Nov 15 10:05 res_odbc.conf |
| 10:13.07 | Telemac | [ast_cnf] |
| 10:13.07 | Telemac | enabled => yes |
| 10:13.07 | Telemac | dns => asterisk |
| 10:13.07 | Telemac | username => xxx |
| 10:13.07 | Telemac | password => yyy |
| 10:13.07 | Telemac | pre-connect => yes |
| 10:18.34 | *** join/#asterisk alainr (n=Alain@vpn-wh.rz-zw.fh-kl.de) |
| 10:18.38 | alainr | hi |
| 10:19.07 | alainr | anybody knows if asterisk supports sha encrypted passwords in sip.conf? |
| 10:19.37 | BBHoss | i think just md5 |
| 10:19.43 | billybongo | dns => asterisk should be |
| 10:19.44 | billybongo | dsn => asterisk |
| 10:20.41 | billybongo | Telemac: ^^ |
| 10:20.54 | alainr | hm, thats bad... |
| 10:21.57 | BBHoss | why is that 'bad' the majority of web apps use md5 hashed passwords? They may be somewhat insecure, but they're better than plaintext! |
| 10:22.27 | alainr | true, but i have ldap and it stores in sha |
| 10:22.35 | BBHoss | oh |
| 10:22.43 | BBHoss | heh, i see your predicament |
| 10:23.22 | billybongo | alainr: user openser and ldap users? |
| 10:24.09 | Telemac | billybongo: Arrr, thanx. Finally I should have paste it rather writing it by hand, or should by new eyes. (Should asterisk indicates an parse error about that ?) |
| 10:24.33 | BBHoss | IAX2 uses RSA auth |
| 10:24.39 | *** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60) |
| 10:24.57 | BBHoss | also the reason MD5 is the only supported type is because thats whats in the RFC for SIP |
| 10:25.07 | billybongo | Telemac: probably asterisk should complain - surprised that it didn't |
| 10:25.13 | BBHoss | like digium really follows standards! |
| 10:25.39 | alainr | <billybongo> i have not tried openser, but will have a look at it |
| 10:25.39 | Telemac | billybongo: Haven't any error |
| 10:26.50 | billybongo | alainr: there are some howtos on using ser/openser for registering and * for other stuff - could be the quickest route if you want to use ldap |
| 10:26.51 | BBHoss | even if you got asterisk to support SHA hashes, you would have to have a phone that supported SHA |
| 10:28.27 | alainr | <BBHoss> no, phone sends clear text to asterisk and asterisk authentifies with the sha password in sip.conf |
| 10:28.53 | BBHoss | well that would be doable, but you'd need to write a patch |
| 10:29.58 | BBHoss | why dont you just put the sha hash and the phones password the same thing? |
| 10:30.38 | alainr | ? |
| 10:31.04 | alainr | you can't concert sha to clear text |
| 10:31.21 | alainr | onvert |
| 10:31.55 | alainr | and if user changes his password, the sip password should also be changed |
| 10:32.07 | BBHoss | ol |
| 10:32.10 | BBHoss | ok |
| 10:33.09 | JT | alainr: since when do phones auth with clear text? |
| 10:34.42 | *** join/#asterisk anujsingh (n=anuj@59.94.141.178) |
| 10:34.49 | anujsingh | Hii |
| 10:35.16 | alainr | don't they? if i tell them to do so |
| 10:35.18 | anujsingh | how to configure loadbalancing between two asterisk servers? |
| 10:35.45 | JT | alainr: not normally |
| 10:36.19 | anujsingh | can i have some tutorial howto ? for load balancing between two asterisk servers ? |
| 10:36.24 | anujsingh | thanks in advance. |
| 10:36.54 | JT | anujsingh: openser perhaps |
| 10:39.33 | billybongo | anujsingh: you have two options, use SRV DNS records to send users to one or other asterisk or configure ser/openser to sit on the front of your pair of asterisks |
| 10:40.59 | JT | well there are more options than that ;) |
| 10:41.07 | anujsingh | yes? |
| 10:41.29 | JT | Linux HA |
| 10:41.33 | anujsingh | i am using vicidial , heartbeat and drbd |
| 10:41.37 | JT | having a group of extensions each |
| 10:41.55 | anujsingh | common mysql database |
| 10:42.09 | anujsingh | replicating with drbd |
| 10:42.30 | JT | heh mysql |
| 10:45.17 | billybongo | anujsingh: OK, there are an infinity of options, of which the 2 I gave was a small subset ;-) |
| 10:45.57 | anujsingh | which one is simple ? i am n00b with aster |
| 10:45.58 | billybongo | anujsingh: what are you trying to achieve? Complete scalability/redundancy ? |
| 10:47.38 | anujsingh | yes scalability/redundancy |
| 10:48.31 | JT | anujsingh: that's not a simple goal |
| 10:49.57 | anujsingh | without trying nothing can be done. I want to try |
| 10:50.24 | anujsingh | i saw someone showing balncing with vicidial monitor tool ,. |
| 10:50.42 | JT | sure |
| 10:50.56 | anujsingh | such as above 15 calls next call will be transfered to anothe aster. |
| 10:50.56 | JT | i'm just saying that the wish for it to be simple is unreasonable |
| 10:52.35 | billybongo | anujsingh: if I was doing this I would have a load-balanced openser cluster on the front end, as many asterisks as you need to handle the media, and a database cluster of your choice to hold the config/voicemails etc |
| 10:52.36 | anujsingh | is it possible to use balance package for load balancing ? |
| 10:53.23 | JT | balance package? |
| 10:53.29 | anujsingh | someone claming he has done it with balance . |
| 10:53.31 | billybongo | anujsingh: do your asterisk users need to find each other across different machines? |
| 10:53.31 | anujsingh | yes |
| 10:54.10 | billybongo | if so you have to think it out very carefully |
| 10:54.36 | anujsingh | neh both the servers will be at a single place. |
| 10:54.42 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 10:54.46 | anujsingh | same users. |
| 10:55.04 | anujsingh | same network same users. |
| 10:55.21 | alainr | JT is it then not possible to force the phone to send cleartext or is this against sip standard? |
| 10:55.22 | billybongo | ok |
| 10:55.47 | billybongo | anujsingh: so if user A is registered with server 1 and user B is registered with server 2, how will they find each other? |
| 10:56.42 | JT | alainr: pretty sure it's not possible |
| 10:56.46 | JT | alainr: it's also stupid |
| 10:59.55 | *** join/#asterisk Steven_elvisda (n=Steven_E@202.47.107.60) |
| 11:00.56 | alainr | JT is there a problem if i would use md5 in sip.conf, does it work with all phones? or is it recommended to use cleartext in sip.conf |
| 11:19.10 | *** join/#asterisk _ys (i=ys@91.151.196.254) |
| 11:19.19 | JT | alainr: just use cleartext, asterisk will make the md5 hash on demand |
| 11:20.15 | alainr | so using md5 is also not good? |
| 11:20.22 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 11:20.27 | JT | i guess it's fine |
| 11:20.33 | JT | plaintext is more convenient |
| 11:20.38 | JT | over sip it's MD5 anyway |
| 11:20.49 | alainr | i don't like to store any password in cleartext |
| 11:22.08 | *** join/#asterisk ussrback (i=MAX@87.253.33.236) |
| 11:22.58 | *** join/#asterisk s0lid (n=_freq@211.243.50.60.brf03-home.tm.net.my) |
| 11:23.00 | alainr | so if sip is md5 anyway i just store it as md5 |
| 11:23.10 | JT | i guess you can |
| 11:23.49 | alainr | but still have to find a way to get password from ldap sha to md5 |
| 11:24.08 | ussrback | hi all |
| 11:24.12 | alainr | maybe with storing it double when registering users |
| 11:24.19 | JT | alainr: that's impossible |
| 11:24.23 | JT | i guess you'll have to |
| 11:24.41 | alainr | conversion is impossible, i know |
| 11:25.05 | alainr | does asterisk work fine with radius? |
| 11:28.28 | *** join/#asterisk Tiffon (n=name@unaffiliated/tiff0n) |
| 11:29.18 | ussrback | what should i install to use H323 ? |
| 11:29.32 | ussrback | is it necessary to install Openh323 |
| 11:29.34 | ussrback | ? |
| 11:29.40 | JT | preferably not asterisk ;) |
| 11:30.05 | tzafrir | ussrback, no. you can also use ooh323c from addons |
| 11:30.35 | tzafrir | and IIRC you had a problem building it yesterday, and I recall asking you for a more complete trace |
| 11:30.42 | ussrback | tzafrir: i just need to compile this addon and put so in my modules dir? |
| 11:30.49 | JT | chan_woomera is meant to be the best H.323 option |
| 11:31.03 | tzafrir | 'make install' should do that |
| 11:31.13 | ussrback | JT: why woomera is the best solution? |
| 11:31.29 | JT | ussrback: the rest are very unreliable |
| 11:32.25 | ussrback | but shoud i need to install woomera on separate box? |
| 11:32.44 | JT | no, the box does not need to be seperate |
| 11:33.05 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 11:33.18 | JT | ussrback: btw, what are you connecting to H.323 for? phones, provider? |
| 11:34.56 | ussrback | provider |
| 11:35.11 | ussrback | ill send international calls throught the h323 |
| 11:35.16 | ussrback | and receive them |
| 11:35.29 | JT | if you don't have luck with chan_woomera you may just want to use Yate as a H.323 <---> SIP gateway |
| 11:35.36 | ussrback | on the other side ill have sip users |
| 11:36.11 | ussrback | where from can i download woomera? |
| 11:44.52 | *** join/#asterisk coppice (n=chatzill@102.204.17.210.dyn.pacific.net.hk) |
| 11:44.56 | *** join/#asterisk ai-a (n=jake2@cpc5-oxfd1-0-0-cust523.oxfd.cable.ntl.com) |
| 11:47.05 | ai-a | can i use the manager port to track call transfers (attended) calls ? and who out of all Dial(SIP/,....) finally picks up the call ? I want to track all calls as they flow via transfers, and even parked, from ringing to termination. |
| 11:48.21 | *** join/#asterisk bantu (n=Miranda@p54A32965.dip0.t-ipconnect.de) |
| 11:50.53 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 11:52.12 | *** join/#asterisk elzapp (n=base@unclesam.elzapp.com) |
| 11:59.06 | *** join/#asterisk ussrback (i=MAX@87.253.33.236) |
| 12:20.56 | *** join/#asterisk Pondiboy (n=Pondiboy@122.164.124.59) |
| 12:26.52 | *** join/#asterisk Haris (n=Haris@unaffiliated/haris) |
| 12:26.54 | Haris | Hello guys |
| 12:27.01 | Pondiboy | hi |
| 12:27.28 | Haris | Can we hook up normal pstn phone sets to Avaya IP office digital lines and talk through them or do we absolutely need an IP/SIP phone to talk on the digital lines? |
| 12:28.25 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 12:28.54 | Pondiboy | Hmm donno i am a newbie there m8 |
| 12:34.14 | tzafrir | Haris, how exactly is this a question about Asterisk? |
| 12:34.43 | cpina | hello everybody |
| 12:34.47 | tzafrir | hi |
| 12:34.51 | tzafrir | what's up? |
| 12:35.30 | Haris | tzafrir: Its not! |
| 12:37.27 | *** join/#asterisk cpina (n=carles@ip23498.bcn.altecom.net) |
| 12:37.38 | cpina | tzafrir: 1 moment... |
| 12:38.02 | *** join/#asterisk b1ch0 (n=ralabiso@static-200-105-209-46.acelerate.net) |
| 12:38.05 | tzafrir | Haris, you can't simply hook them |
| 12:38.33 | Haris | So that means, only sip phones can be used with it |
| 12:38.49 | Haris | but normal phone sets could still be used with analog lines in it? |
| 12:41.51 | cpina | tzafrir: about my ztdummy |
| 12:41.54 | cpina | :-) |
| 12:42.03 | cpina | we changed the server while we buy a hardware card :-) |
| 12:43.36 | tzafrir | Haris, yes, you need some sort of analog adapter for analog phones |
| 12:43.38 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 12:43.49 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 12:44.43 | *** join/#asterisk michael-i (n=michael-@Lc371.l.pppool.de) |
| 12:46.01 | ai-a | If the E1 card is Alaw, and the phone is Alaw, does asterisk just forward the audio stream? we seem to be having some sort of 'auto gain' effects on our calls after moving to Asterisk from old hardware pbx. what could be causing this? |
| 12:46.29 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 12:49.36 | *** join/#asterisk penguinFunk (n=penguin@unaffiliated/penguinfunk) |
| 12:51.34 | JT | ai-a: the fact that your handsets are different? |
| 12:52.21 | *** join/#asterisk killfill (n=killfill@pc-164-134-45-190.cm.vtr.net) |
| 12:52.28 | killfill | hi.. |
| 12:52.34 | killfill | in show channels, i see this: Local/65@default-921 65@default:4 Ring TrySystem(fetch -T 2 "http://l |
| 12:52.43 | b1ch0 | hi everibody, i am planning to use G729 (0r 723) on a new installation because of all my phones support those codecs, what append if i enable an IVR (with "729 voices) ? do i have to pay any kind of licence ? |
| 12:52.47 | killfill | its stuck in there. how would i kill that channel? |
| 12:53.00 | killfill | asterisk it at 100%cpu.. i think becouse of that... |
| 12:55.06 | ai-a | JT: its a call from outside coming into the building and answered by a snom300 phone. |
| 12:55.33 | ai-a | asterisk records the call, and we can hear that when they are not talking the audio is louder. |
| 12:56.29 | b1ch0 | or if i need to listen my voicemail ? |
| 13:01.36 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 13:03.56 | cpina | in one asterisk server, after type show sip channels, there is a lot of BYE state channels, with unknown codec |
| 13:04.02 | cpina | how we could "remove" or destroy it? |
| 13:08.20 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:09.07 | JT | ai-a: did your old pbx use Snom 300 phones? |
| 13:09.54 | tzafrir | killfill, soft hangup ? |
| 13:10.00 | tzafrir | kill the processes? |
| 13:10.05 | *** part/#asterisk Haris (n=Haris@unaffiliated/haris) |
| 13:11.08 | tzafrir | (pkill, or ps auxww | grep whatever | grep -v grep | awk '{print $2}' |xargs kill |
| 13:11.08 | tzafrir | ) |
| 13:11.08 | cpina | :-) |
| 13:11.08 | cpina | i mean sip channels |
| 13:11.18 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 13:11.22 | cpina | IP_ADDRESS xxxxx yyyyyyy 00102/00103 unkn No (d) Rx: BYE |
| 13:11.27 | cpina | we have had 180 of it |
| 13:11.41 | cpina | yes, it has flag "d" to destroy |
| 13:11.44 | cpina | i would like to destroy before |
| 13:11.45 | cpina | :-) |
| 13:12.06 | cpina | tzafrir: ops, you was answering to killfill not to me :-D |
| 13:12.26 | ai-a | JT: nope. |
| 13:12.45 | tzafrir | cpina, I suppose you have tried a soft hangup, right? |
| 13:12.55 | cpina | tzafrir: is only for active channels, yes |
| 13:12.56 | *** join/#asterisk El_Capitan (n=BluesBoy@217.6.11.154) |
| 13:13.02 | ai-a | old system was completly different. but using the same E1 line. we've added asterisk pbx, and 50+ snoms. |
| 13:13.27 | ai-a | i cant see any auto gain in the snom or sangoma card settings. So wondering if asterisk is performing auto gain. |
| 13:14.04 | ai-a | come to think of it.,. you can hear the auto gaining in the recordings which have nothing to do with the snom, so it must be either the asterisk, sangoma or e1 connection somewhere performing this auto-gain |
| 13:14.06 | *** join/#asterisk mosty (i=mostyn@60-241-198-194.static.tpgi.com.au) |
| 13:15.53 | JT | ai-a: i think you're being slightly unreasonable expecting the volumes to be completely the same |
| 13:16.01 | mosty | i want to accept a call on one asterisk server, and terminate it via a second asterisk server, such that the billsec in the cdr entries is the same on the two servers- is this possible? |
| 13:16.18 | ai-a | JT: when the customer isnt talking the volume goes up and we hear a lot of background noises. |
| 13:16.29 | *** join/#asterisk rob0 (i=rob0@sorry.nodns4.us) |
| 13:16.41 | ai-a | Telesales are saying this makes it hard to hear the customer as it seems to auto-gain too quickly. |
| 13:16.49 | JT | ai-a: less silence supression, or clearer phones, take your pick |
| 13:16.51 | ai-a | so they turn the vol down, but then the customer is quiet. |
| 13:16.57 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:17.02 | ai-a | JT: what does auto-gain ? |
| 13:17.08 | JT | nothing in asterisk |
| 13:17.17 | ai-a | exactly what i thought. |
| 13:17.31 | ai-a | and the sangoma experts say the card doesnt do auto-gain either. |
| 13:17.44 | JT | what card is it |
| 13:17.48 | ai-a | maybe its the customer's analogue phone that is auto gaining for them. |
| 13:17.52 | ai-a | A101D |
| 13:18.02 | JT | no, it's probably not auto gain at all |
| 13:18.17 | JT | probably the old pbx supressed background noise |
| 13:18.32 | ai-a | I see. |
| 13:19.19 | JT | the only "auto-gain" possibility i see is in the snoms |
| 13:19.23 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:19.23 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 13:19.25 | JT | tried a different phone? |
| 13:19.46 | ai-a | nope. but if its the phone, asterisk monitor wouldnt notice it. |
| 13:20.08 | JT | just try a few different phones |
| 13:20.20 | *** join/#asterisk Lasse123 (n=Lasse@int-gw.algitech.com) |
| 13:21.07 | ai-a | thing is, we have 20+ guys, and 1% complaints... but the bosses want 100% perfection. |
| 13:21.42 | JT | so do some tests for them? :) |
| 13:22.06 | ai-a | i prefer them to shut up and be happy with the new tech.. they are lo paid :) |
| 13:22.07 | ai-a | *low |
| 13:22.14 | JT | heh |
| 13:22.53 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 13:23.48 | lirakis | Does anyone know of good specific resources for ISDN signaling functionality in asterisk? |
| 13:24.03 | JT | err, what? |
| 13:24.07 | lirakis | im curious how asterisk responds to certain codews |
| 13:24.09 | lirakis | *codes |
| 13:24.10 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 13:24.35 | JT | then the best documentation is the source |
| 13:24.59 | lirakis | JT: .. like can i configure asterisk to send 34 when i get a specific isdn code |
| 13:25.25 | *** join/#asterisk Telemac (n=telemac@213.223.113.74) |
| 13:25.28 | JT | probably by modifying chan_zap.c or similar |
| 13:25.31 | Telemac | Hello (again) |
| 13:29.34 | Telemac | I'm still having some trouble to setup realtime with odbc. dsn is properly resolved and connected by asterisk (odbc show), when I execute "SELECT * FROM sip_conf WHERE name='x'" within isql I get matching row, but when I check it within asterisk with "realtime load sipusers name x" nothing is found. In extconfig.conf sipusers is bound to "odbc.ast_conf.sip_conf", ast_conf is my working dsn and sip_conf is the table name. What's wrong there ? |
| 13:33.06 | *** join/#asterisk sasch (n=info@host117-234-static.4-79-b.business.telecomitalia.it) |
| 13:33.08 | sasch | hi all |
| 13:33.11 | *** join/#asterisk ming_zym (n=ming_zym@124.14.233.147) |
| 13:34.02 | sasch | anyone know a link for echo cancellation with tdm400p ?? |
| 13:34.25 | [TK]D-Fender | sasch: All you have is Zaptel. Go WIKI it up. |
| 13:35.22 | sasch | can you get me a link |
| 13:35.23 | [TK]D-Fender | sasch: so taht being the routines that come with Zaptel as a base, HPEC from Digium (go to their site to check out), or OLSEC ( http://www.rowetel.com/ucasterisk/oslec ) |
| 13:35.29 | [TK]D-Fender | ~wikis |
| 13:35.30 | jbot | [wikis] http://www.voip-info.org |
| 13:35.33 | sasch | excusme for my english I'm italian :-P |
| 13:35.56 | sasch | thanks !!! |
| 13:37.31 | Lasse123 | BUG? Incoming PSTN call to asterisk with prefered codec ALAW (but iLBC supported), asterisk forwards call to client which responds "200 OK" with sdp indicating it only supports iLBC, asterisk responds with ACK and starts P2P bridging call which results in client receiving ALAW? |
| 13:41.35 | mosty | is it possible to get asterisk to optimize the choice of codec in order to avoid transcoding? ie if i'm trying to forward on a call using codec X to another asterisk server which supports X, use that codec even if it's not the first priority (first allow) |
| 13:42.23 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 13:42.42 | puzzled | hi |
| 13:44.26 | [TK]D-Fender | mosty: Yes. Make another peer with only the codec you want in it. |
| 13:44.33 | robl^ | hrmmm -- ~wikis is plural, yet jbot returns a single wiki. I declare a bug! |
| 13:45.40 | [TK]D-Fender | robl^: ~wiki is a reserved jbot function for Wikipedia |
| 13:45.59 | mosty | [TK]D-Fender, is it possible to route calls based on codec in the dialplan? eg send g729 calls to gateway A, send gsm calls to gateway B |
| 13:46.05 | robl^ | ahhh!! I knew there was a simple answer ;-) |
| 13:46.40 | [TK]D-Fender | mosty: You can get the codec of the current channel easily enough. "show functions" <- |
| 13:49.08 | *** join/#asterisk maruz (n=maumar@gw.cost.it) |
| 13:50.41 | _x86_ | morning all |
| 13:51.53 | Telemac | I've configured extconfig.conf but it looks like this file was not loaded. Is it read by default or should I ensure some module to be loaded (res_config ?) |
| 13:51.57 | mosty | [TK]D-Fender, do you mean CHANNEL? i think that's new in 1.4, and i need to use 1.2 |
| 13:52.17 | *** join/#asterisk duckz (n=duckz@85.204.47.228) |
| 13:52.20 | _x86_ | mosty: 1.2 isn't supported anymore |
| 13:52.55 | mosty | i'm stuck with 1.2 until there's a viable backport for debian etch, unfortunately |
| 13:52.58 | [TK]D-Fender | mosty: SIPCHANINFO(item) Gets the specified SIP parameter from the current channel <-------- |
| 13:53.18 | mosty | i saw SIPCHANINFO, but i need this to work with IAX |
| 13:53.25 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 13:53.28 | [TK]D-Fender | mosty: Thinking you're restricted to friggen PACKAGES. How sad... |
| 13:53.37 | _x86_ | mosty: dude, compile Asterisk from source... it's easy |
| 13:53.48 | _x86_ | mosty: pre-compiled asterisk is utter crap |
| 13:53.56 | *** join/#asterisk markit (n=marco@88-149-177-66.static.ngi.it) |
| 13:54.01 | [TK]D-Fender | mosty: IAXPEER(<peername|CURRENTCHANNEL>[:item] <------- |
| 13:54.07 | mosty | i can compile asterisk from source OK, but i run too many machines for that to be practical |
| 13:54.27 | [TK]D-Fender | mosty: and you upgrade * how often? |
| 13:54.39 | markit | hi, I've heard that ztdummy has a recent bug... has it been solved in svn stable? someone with a bug # to give me? a friend of mine has problems with conference and meetme... |
| 13:54.40 | _x86_ | mosty: compile it once the way you want it, and make a tarball of the output... copy from machine to machine... |
| 13:54.49 | ZenBSDi | not really mosty.. just learn to compile and create your own .deb pkgs so you can just dpkg install it =p |
| 13:54.57 | [TK]D-Fender | mosty: Either way, I've just given you the answer |
| 13:55.13 | *** join/#asterisk shinao1 (n=shinao1@209.159.162.105) |
| 13:55.22 | mosty | actually i am learning to build asterisk deb's, but i'm not yet confident enough that my packages are ok |
| 13:55.43 | mosty | [TK]D-Fender, thanks, i will see what i can get done given my constraints |
| 13:57.17 | ZenBSDi | another bit of advice mosty.. seeing as how debian's developers move at a pace only seen by turtles.. you might consider switching to ubuntu server going forward so you aren't stuck waiting for those molasis asses to get things into gear =p |
| 13:57.43 | ZenBSDi | oh and if you know any deb devels.. please quote me verbatum =p |
| 13:58.10 | mosty | ZenBSDi, i actually like the pace of debian for 99% of the packages |
| 13:58.33 | ZenBSDi | if you *like* the pace of debian .. you'd *love* the pace of ubuntu =p |
| 13:59.06 | ZenBSDi | thats why I gave up debian.. the U-crew is tearing it up!!! |
| 13:59.10 | mosty | i don't want anything to change unless it has to. which is good for most things, but not something that's still maturing like asterisk unfortunately |
| 13:59.41 | mosty | i could try ubuntu lts i suppose, what version of asterisk does that have? |
| 14:00.19 | ZenBSDi | asterisk is very mature as it sits despite what even the devels might say.. I just put an asterisk system into place and here it is .. 600+ uses already .. p4 3ghz with 1 gig of ram.. this thing is smoking with 30 users each |
| 14:00.32 | ZenBSDi | based on 1.4.10 |
| 14:00.45 | ZenBSDi | I've got some java scripts using the fastagi interface |
| 14:00.56 | ZenBSDi | load is sitting pretty at 1.6 .. |
| 14:01.04 | ZenBSDi | oh .. it's ubuntu 7.10 server too |
| 14:01.05 | ZenBSDi | heh |
| 14:01.50 | mosty | regular ubuntu releases are not stable enough for me |
| 14:02.03 | ZenBSDi | how long you been a linux user? |
| 14:02.19 | mosty | more than 10 years, i don't remember exactly |
| 14:03.04 | mosty | if this was just a small asterisk box like yours it would not be a problem, but i admin many asterisk boxes, and some of them are large |
| 14:03.18 | [TK]D-Fender | ZenBSDi: Know what the worst part of a head-on collision between two turtles is? The hours of screaming :D aaaaaaaaaaaaaaaahhhhhhhhhhhhhhh!!!!!!!!!!!!!!!!!!!!!!!!!!! |
| 14:03.40 | ZenBSDi | puhlease.. you can't possibly have a decade of this crap and be afraid "stability" issues .. incase you haven't noticed even the mighty old and iron forged debian has "stability" issues. if you have a decade of linux .. you'd be able to fix it so you dont' have to deal with the other 80% of things debian does wrong. |
| 14:03.57 | Telemac | Is extconfig.conf supported with asterisk 1.2.21.1-r1 ? |
| 14:04.28 | [TK]D-Fender | ZenBSDi: And turtles are pretty quick too... I'd say more akin to molasses going uphill in January ;) |
| 14:05.24 | ZenBSDi | [TK]D-Fender, colliding turtles.. or going uphill .. either way.. it's more development unfolding quicker than debian devels.. =p |
| 14:05.28 | mosty | ZenBSDi, i only run debian stable on boxes that i do not want to touch if i can avoid it. i don't have the time to keep upgrading these machines when the security support ends for a certain release of something like fedora |
| 14:06.01 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 14:06.11 | [TK]D-Fender | mosty: "yum update" <- CentOS |
| 14:06.50 | mosty | [TK]D-Fender, sure but if you upgrade between releases that can break things easily |
| 14:07.19 | *** join/#asterisk irule (n=irule@200.53.61.4) |
| 14:09.00 | ZenBSDi | who said anything about fedora mosty.. I think you need to read what I said.. I said . and here I'll break down itno parts... Going forward .... so you are NOT stuck waiting for debians development cycles .... you should consider using ubuntu server.. so what I said is. |
| 14:09.29 | PBXX | i have problem with /usr/local/sbin/faxgetty when he is runnign from inittab he cannot red buffer from iaxmodem |
| 14:09.33 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 14:09.37 | PBXX | when i run it on shell everything its fine |
| 14:09.47 | coppice | once you've waited for debian release cycle, continental drift seems really fast :-) |
| 14:10.08 | rob0 | Stop Continental Drift! Sign the petition. |
| 14:10.28 | *** join/#asterisk freezey (n=freezey@maher.mercy.edu) |
| 14:10.32 | ZenBSDi | but enough of the distro war.. this is about asterisk .. asterisk rocks! |
| 14:10.35 | mosty | zenbsdi: i was just exagerating to make a point. i need a distribution with a long support life. regular ubuntu is too frequent for my situation. ubuntu LTS may be ok, i don't know |
| 14:10.43 | *** join/#asterisk Haris (n=Haris@unaffiliated/haris) |
| 14:10.45 | Haris | hello guys |
| 14:10.47 | freezey | if somebody set a password to their extension's voicemail how exactly do i reset the password? |
| 14:10.50 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 14:10.53 | Haris | Is there a difference between a digital phone and a sip phone? |
| 14:11.07 | ZenBSDi | sip is a protocol .. |
| 14:11.27 | mosty | nothing changing means nothing new breaking. my users hate it when their phones don't work |
| 14:11.48 | ZenBSDi | so you should look into a digital phone that supports sip =p |
| 14:11.58 | [TK]D-Fender | Haris: Typically the term "digital" refers to proprietary electron signalling that is tied to a single vendor and won't work with any other system. |
| 14:12.14 | Haris | Avaya 4406D, Avaya 5410 |
| 14:12.28 | Haris | [TK]D-Fender: I see |
| 14:12.31 | ZenBSDi | akkk [TK]D-Fender propietary!!! |
| 14:12.35 | Haris | [TK]D-Fender: Still, is it sip capable? |
| 14:12.39 | [TK]D-Fender | Haris: a SIP phone is a phone that uses TCP/IP and speaks the SIP protocol to communicate with typically pretty much any SIP server. |
| 14:12.48 | Haris | I see |
| 14:12.49 | *** join/#asterisk macros73_ (n=cs@dsl093-063-226.pit1.dsl.speakeasy.net) |
| 14:12.54 | freezey | [TK]D-Fender: if somebody set a password to their extension's voicemail how exactly do i reset the password? |
| 14:13.01 | Haris | so we can say its a normal pstn phone, but a properitery one? |
| 14:13.05 | freezey | [TK]D-Fender: you seem to answer most questions |
| 14:13.29 | [TK]D-Fender | freezey: vi /etc/asterisk/voicmail.conf |
| 14:13.36 | ZenBSDi | [TK]D-Fender is an asterisk demi-god.. so of course he answers the most q's |
| 14:13.53 | Haris | demo god? |
| 14:13.55 | Haris | demi+ |
| 14:14.04 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 14:14.08 | Haris | are they still alive? |
| 14:14.15 | [TK]D-Fender | ZenBSDi: No, I'm not that good, but I know most of the basics of day to day use. I don't do DB's, AGI, or other coding, and suck at linux (mostly) |
| 14:14.37 | *** join/#asterisk ussrback (n=MAX@80.241.177.19) |
| 14:14.42 | [TK]D-Fender | Haris: what does "normal" mean if its "proprietary"? |
| 14:14.49 | Haris | Ok, on the subject of proprietary stuff |
| 14:14.53 | freezey | [TK]D-Fender: ok i see which line you mean 4586111,username,email@mail.com,,|tz=eastern|attach=yes|saycid=no|review=no|operator=no|envelope=yes|delete=no so that 4586111 should be password? |
| 14:15.03 | [TK]D-Fender | freezey: Yes. |
| 14:15.05 | Haris | would normal non-proprietary digital phones work with Avaya IP phone 406? |
| 14:15.10 | Haris | phone = Office |
| 14:15.11 | ZenBSDi | word.. if you start doing the agi coding you'll really own then dude.. agi coding is where you can do that stuff like call in and pay with a CC over the phone.. the script grabs that stuff and processes it |
| 14:15.15 | freezey | [TK]D-Fender: tried usin ghtat passowrd and still got denied |
| 14:15.16 | ZenBSDi | thats what I'm working on now |
| 14:15.39 | [TK]D-Fender | Haris: You can get line cards for Avaya to support standard analog phones. |
| 14:15.50 | Haris | [TK]D-Fender: on digital ports? |
| 14:15.53 | Haris | or sip ports? |
| 14:15.55 | [TK]D-Fender | freezey: And what did you do to APPLY your changes? |
| 14:15.57 | Haris | sip capable+ ports |
| 14:16.04 | freezey | dialplan reload |
| 14:16.27 | _x86_ | [TK]D-Fender: eh? |
| 14:16.37 | [TK]D-Fender | Haris: Depends on your line card. Some mfg's digital line ports ALSO support analog phones directly. In other cases you may need to by a card DEDICATED to that |
| 14:16.39 | _x86_ | [TK]D-Fender: you are teh suck at leenocks? |
| 14:16.51 | [TK]D-Fender | freezey: voicemail != dialplan |
| 14:17.08 | [TK]D-Fender | freezey: "module reload app_voicemail.so" |
| 14:17.10 | *** join/#asterisk axisys (i=iqbala@outbound.silenceisdefeat.org) |
| 14:17.13 | Haris | means, additional! cost |
| 14:17.26 | freezey | [TK]D-Fender: great thanks |
| 14:17.43 | [TK]D-Fender | Haris: One prime reason that proprietary PBX SUCK. Welcome to vendor lock-in, and stock up on KY. |
| 14:18.08 | b1ch0 | JC: hi still there ? |
| 14:18.17 | freezey | [TK]D-Fender: and to reset the mailbox like deleted all the voicemails in their is their a way to do it through asterisk? |
| 14:19.08 | [TK]D-Fender | freezey: to fully reset a box you would delete that box's folder entirely. You can code something in the dialplan for this easily enough. |
| 14:19.08 | b1ch0 | im tring to test transversal nat, and i cant understand if externip=222.222.222.222 is router address or my * natted IP |
| 14:19.25 | freezey | [TK]D-Fender: ty |
| 14:19.26 | [TK]D-Fender | b1ch0: its your routers EXTERNAL IP |
| 14:19.42 | [TK]D-Fender | b1ch0: the address it gets from your provider. |
| 14:19.47 | *** join/#asterisk asdx (n=diego@adsl-147-91.click.com.py) |
| 14:20.07 | b1ch0 | ok, thanks , the same as my * default gateway, right ? |
| 14:20.31 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 14:20.33 | [TK]D-Fender | b1ch0: Dunno what your * gets for that. I gave you the only answer that counts. Run with it. |
| 14:21.10 | [TK]D-Fender | b1ch0: typically systems behind NAT routers get PRIVATE addresses which is NOT what is called for here. |
| 14:21.47 | krdian_ | <PROTECTED> |
| 14:21.47 | Telemac | I'm trying to test realtime from odbc. My dsn seems ok, I've configured extconfig.conf but when I try realtime load in asterisk console no matching row is found. So I wonder extconfig.conf was loaded by default with asterisk 1.2, or even supported ? I'd like to have the whole sip.conf in DB, should I used "switch => ..." statement anywhere or just extconfig.conf ? |
| 14:21.52 | krdian_ | hello |
| 14:22.26 | rob0 | whatismyip.org (which doesn't work if your ISP uses HTTP proxying) |
| 14:22.56 | [TK]D-Fender | Telemac: Real-time requires you to code your contexts into extensions.conf and use the switch statement to tell * to use Realtime. You must do this for ALL contexts. So the CONTEXTS may be dynamic, but not the STRUCTURE. |
| 14:23.25 | [TK]D-Fender | c/CONTEXTS/CONTENTS (extens)/ |
| 14:23.30 | b1ch0 | ok, i understand .... so, after IAX test (that work great behind the transversal nat i got) im going to try if SIP works well or not following http://www.aocomputing.net/?p=3 instructions |
| 14:23.42 | b1ch0 | wish me luck |
| 14:23.53 | [TK]D-Fender | b1ch0: And then we'll see how good you are at following instructions :) |
| 14:24.18 | b1ch0 | :-) |
| 14:24.25 | Telemac | [TK]D-Fender: Ok, thanx I will look at that |
| 14:26.40 | mosty | [TK]D-Fender, it appears that ${SIPPEER(account:codec[0])} shows the prefs in sip.conf, but what i need is only those prefs in sip.conf that the caller supports. is it possible to figure that out? |
| 14:27.06 | [TK]D-Fender | mosty: Dunno. |
| 14:27.16 | shinao1 | hi.. has anyone ever used elastix before? with xorcom astribanks? |
| 14:29.34 | tzanger | coppice: (or anyone, really) -- any good links or references for fixed-point echo cancellation for short spans? |
| 14:30.07 | coppice | what do you mean by a short span? |
| 14:30.14 | Qwell | one that isn't long? |
| 14:30.18 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 14:30.38 | [TK]D-Fender | What about medium spans? Or medium + smidgeon? |
| 14:30.57 | tzanger | yeah, a short-length span. sliptest is reporting between 700-900 for the span |
| 14:31.21 | tzanger | that's not ms, but I'm guessing it's frames? (125us?) |
| 14:32.17 | coppice | have you tried OSLEC? |
| 14:32.40 | tzanger | no, I haven't looked at it yet, it's on my list though |
| 14:33.25 | krdian_ | huh, anybody had problem with Astrisk::AGI 0.10 ? |
| 14:33.32 | tzanger | ahh it's already targeted for blackfin |
| 14:33.47 | coppice | duh! of course |
| 14:33.53 | tzanger | I didn't realize oslec was rowetel |
| 14:34.09 | b1ch0 | ... |
| 14:34.52 | b1ch0 | hi, how about configuring all internal phones using IAX instead of SIP in an internal LAN ? |
| 14:35.23 | Telemac | Still trying to setup odbc realtime. My dsn is ok (loaded/connected in asterisk), I've completed extconfig.conf, added "switch => Realtime" in extensions.conf as first for "[mycontext]" but it still fails to get matching row in asterisk console with "realtime load sipusers name user_name"... What could be wrong so ? |
| 14:35.32 | ussrback | when im trying to make install asterisk-addons i got error http://pastebin.ca/775150. how can i fix it? |
| 14:36.23 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 14:36.24 | coppice | hey! Asterisk on an FPGA :-) |
| 14:36.50 | tzanger | :-) |
| 14:37.40 | tzanger | hmm oslec looks interesting for sure |
| 14:37.47 | [TK]D-Fender | b1ch0: What kind of phones? And whats the point of IAX phone INSIDE a LAN? |
| 14:39.21 | puzzled | ussrback, the Makefile has a typo. if you go to the .libs directory you can find the stuff in there |
| 14:39.50 | ussrback | yes i can find |
| 14:40.05 | ussrback | it means that makefile have error? |
| 14:40.18 | *** join/#asterisk zerohalo (n=zeroHalo@pool-71-255-162-167.bstnma.east.verizon.net) |
| 14:41.48 | puzzled | ussrback, yes |
| 14:42.40 | ussrback | ok so how can i install ooh323c |
| 14:42.48 | De_Mon | fix the typo |
| 14:42.51 | *** join/#asterisk yannj_fr (n=yannj_fr@APuteaux-152-1-61-178.w82-120.abo.wanadoo.fr) |
| 14:42.58 | shinao1 | tzafrir: hey |
| 14:43.03 | tzafrir | coppice, I noticed OSLEC has a problem with amd64 here (causes silence). At least the variant in the Debian packages. |
| 14:43.04 | shinao1 | can we still talk some more? |
| 14:43.11 | tzafrir | Haven't had time to invastigate |
| 14:43.30 | shinao1 | how did you do it? i think im using elastix -0.8 |
| 14:43.51 | *** join/#asterisk Darthclue (n=e054502@fw149.nisd.net) |
| 14:43.59 | tzanger | hmm, I would have thought NAPI would make TDMoE worse, not better |
| 14:44.11 | tzanger | but that's just theorizing, I haven't tested yet |
| 14:44.21 | muiro | hey, is there an app to say a currency amount? |
| 14:44.36 | coppice | tzafrir: I haven't tried it, but I understand that works OK. are you sure the version is up to date? I remember there were leak fixes after initial release, but there might be others |
| 14:44.38 | ussrback | how? i am not programmer |
| 14:45.20 | tzafrir | It's a pretty recent one |
| 14:45.33 | puzzled | ussrback, change libchan_h323.so.1.0.1,libchan_h323.1.0.1 in the Makefile and try again |
| 14:46.07 | [TK]D-Fender | muiro: Not in 1 step. You'll have to break it up yourself. |
| 14:46.15 | muiro | ah, ok |
| 14:46.39 | muiro | I'll do it and the contribute it as an addon because that seems like it should be pretty ubiquitous |
| 14:48.23 | *** join/#asterisk irule (n=irule@200.53.61.4) |
| 14:48.25 | [TK]D-Fender | muiro: Too specialized and not hard to di in relatively pure dialplan. |
| 14:48.54 | [TK]D-Fender | muiro: This is much like I class the old "blacklist" stuff.... waste of time. We have existing stuff that con do this easy. |
| 14:49.08 | muiro | yeah, I suppose |
| 14:49.39 | [TK]D-Fender | muiro: What I WOULD advise if you do it in standard dialplan and contribute the code to the WIKI instead. This will both give people the "answer", as well as possibly teach/inspire someone about programming. |
| 14:50.01 | muiro | sounds good |
| 14:50.41 | muiro | I actually haven't gotten into learning how to incorporate either system applications or anything with the AGI yet so it'll be pure dialplan |
| 14:54.36 | *** join/#asterisk af_ (n=getsmart@88-149-240-124.dynamic.ngi.it) |
| 14:54.49 | ussrback | which is config file for chan_ooh323.so? |
| 14:55.14 | [TK]D-Fender | ussrback: ... |
| 14:55.18 | [TK]D-Fender | ~book |
| 14:55.18 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 14:55.20 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.149) |
| 14:55.20 | [TK]D-Fender | ^^^^^^^ |
| 14:56.00 | muiro | it's a good book |
| 14:58.11 | b1ch0 | well it doesnt make much sense installing iax in an inside lan, but that was just an idea (to avoid future provisioning issues) ... just wanted gurus opinion |
| 14:58.34 | [TK]D-Fender | b1ch0: Unless you NEED IAX I advise against its use. |
| 14:58.54 | blitzrage | I prefer SIP myself |
| 14:59.25 | [TK]D-Fender | Yay, Bell just got the HTC Touch.... time to upgrade & grab their 7$ unlim data plan :) |
| 14:59.55 | blitzrage | [TK]D-Fender: huh? |
| 15:00.06 | [TK]D-Fender | blitzrage: http://www.bell.ca/shopping/fr_CA_QC/66393.details?tab=SPECS&colourId=undefined&contractId=term36m |
| 15:00.36 | *** join/#asterisk CunningPike (n=arodgers@204.239.8.149) |
| 15:00.40 | rob0 | On IAX v. SIP, I have 2 sites, an * server at each. I linked them using IAX2, thinking that would be best. Would there be any advantage in changing that to SIP? |
| 15:00.47 | *** join/#asterisk Darthclue (n=chatzill@fw149.nisd.net) |
| 15:00.52 | blitzrage | Bell has a $7 unlimited data plan? |
| 15:01.00 | *** join/#asterisk waverly360 (n=waverly@42.sub-75-200-219.myvzw.com) |
| 15:01.08 | *** part/#asterisk waverly360 (n=waverly@42.sub-75-200-219.myvzw.com) |
| 15:01.18 | [TK]D-Fender | blitzrage: yup, for non-tethered use |
| 15:01.30 | blitzrage | rob0: does it work? If so... why switch? If it ain't broke, don't fix it -- if you don't know why there would be an advantage (or why you are at a disadvantage), then why change? |
| 15:01.42 | coppice | [TK]D-Fender: just got it? i thought it was about to be discontnued |
| 15:01.58 | blitzrage | [TK]D-Fender: hrmmmmmm.... Roger's data plans are RIDICULOUSLY expensive |
| 15:02.00 | rob0 | yeah, I was just curious, given your comment and [TK]D-Fender's. |
| 15:02.03 | [TK]D-Fender | coppice: I know, silly North Americans.... |
| 15:02.36 | [TK]D-Fender | rob0: If it ain't broke... but that is specifically to like 2 * systems, which is the only thing I'd use it for, and only that. |
| 15:03.21 | *** join/#asterisk PodMan99a (n=keith@host81-149-176-8.in-addr.btopenworld.com) |
| 15:03.26 | rob0 | gotcha, ty |
| 15:04.19 | Telemac | I've "[settings]\ sippeers => odbc.ast_cnf.sip_conf" in my extconfig.conf, so what could prevent "realtime load sipusers name _user" from finding matching row within Asterisk console ? |
| 15:04.28 | coppice | [TK]D-Fender: the HTC Touch is cute, and fairly cheap for an HTC phone |
| 15:04.31 | rob0 | Both sites use SIP for external services (orig./term.), and one site has SIP clients. |
| 15:04.32 | *** join/#asterisk truz_`24 (n=truz_`24@74-129-166-232.dhcp.insightbb.com) |
| 15:04.36 | [TK]D-Fender | rob0: Oh... and for use where SIP gets blocked by asshole ISP's |
| 15:04.41 | rob0 | haha yeah |
| 15:05.20 | [TK]D-Fender | coppice: yeah, its base is 2/3 that of the HTC 6800, slimmer, no WIFI, but with that data plan I don't really care. |
| 15:05.31 | coppice | use SIP++ |
| 15:05.36 | coppice | repeating ++ until you find a port they don't piss about with |
| 15:06.02 | PodMan99a | hey all ... have an issue with my asterisk server how can i get my inbound number to dial a queue then have it dial agents .... currently the call enters queue like -- exten => 05601048894,1,Queue(example_queue) but does not get agents called |
| 15:06.03 | coppice | [TK]D-Fender: how much to they charge for it? |
| 15:06.34 | *** join/#asterisk e` (n=e@38.102.196.202) |
| 15:06.35 | [TK]D-Fender | coppice: Here non-subsidized : Touch = $400CAD, 6800 = $600$CAD |
| 15:06.37 | mosty | PodMan99a, how many agents are in the example_queue ? |
| 15:07.11 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 15:07.39 | PodMan99a | 2 |
| 15:07.49 | coppice | [TK]D-Fender: its about US$280 here, last time I looked |
| 15:08.03 | mosty | PodMan99a, and they're both idle when you do "show queue example_queue" ? |
| 15:08.10 | PodMan99a | yes |
| 15:08.11 | [TK]D-Fender | coppice: You know vendor lock-in... OH, and this is the CDMA version. |
| 15:08.32 | coppice | we only see the GSM one |
| 15:08.37 | mosty | PodMan99a, what does it say about callers in the queue? |
| 15:08.43 | PodMan99a | 2 seconds |
| 15:09.03 | PodMan99a | <PROTECTED> |
| 15:09.10 | mosty | bingo |
| 15:09.13 | PodMan99a | phone is logged in and connected as can make calls |
| 15:09.17 | PodMan99a | just im not an agent |
| 15:09.25 | mosty | you haven't logged in as an agent |
| 15:09.30 | [TK]D-Fender | coppice: Yeah, but thats what my cell-co uses and the plan I'm on is really good for here so there's a price to pay... |
| 15:09.52 | [TK]D-Fender | PodMan99a: Not logged in = don't expect to be called by the queue. |
| 15:10.01 | [TK]D-Fender | PodMan99a: Checken & egg show you're running there.... |
| 15:10.07 | [TK]D-Fender | chicken* |
| 15:10.18 | PodMan99a | [TK]D-Fender i am logged in |
| 15:10.19 | coppice | they closed down the CDMA network here. now I've seen a fresh tender thing to start one up again. weird |
| 15:11.10 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 15:11.15 | _x86_ | what's the cheapest CSU that adtran makes? |
| 15:11.19 | _x86_ | single-port |
| 15:11.22 | asdx | lol this dumb-asshole guy (my customer) wants me to help him to build a service provider and he wants to pay me $50 usd. |
| 15:11.25 | asdx | wtf |
| 15:12.14 | PodMan99a | can i paste someone some lines of my config for the queue? |
| 15:12.25 | Darthclue | asdx, tell him that's a per connection charge and charge him everytime that one of his customer's connects |
| 15:12.27 | [TK]D-Fender | asdx: Ask him if he wants fries with that :p |
| 15:12.37 | Darthclue | PodMan99a, use pastebin |
| 15:12.42 | Darthclue | ~pastebin |
| 15:12.43 | jbot | rumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 15:12.48 | PodMan99a | good man Darthclue |
| 15:13.36 | asdx | Darthclue: lol |
| 15:13.39 | asdx | [TK]D-Fender: lol |
| 15:15.14 | PodMan99a | ALL: http://pastebin.com/m5de47094 any ideas ... i can see something is wrong but not sure what |
| 15:15.17 | asdx | i'll do it anyway to learn, he will buy a teliax account so i can play it and learn, then when i have everything up and running i can ask for some cash, and if he doesn't pay i'll just make uninstall asterisk. |
| 15:15.33 | asdx | or turn off the server |
| 15:15.38 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:15.38 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:16.01 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 15:16.33 | Darthclue | Just route all the calls to a recording that says "I'm sorry. I've chosen to be cheap and not pay the developer. To have your service restored, please call me at (insert his cell, home, and other numbers here)." |
| 15:17.22 | asdx | Darthclue: lol, that would be nice as well... |
| 15:17.26 | PodMan99a | Darthclue : I cant be a developer unless i play ... trial and error... have to learn somewhere |
| 15:19.40 | Darthclue | PodMan99a, that was directed at asdx. If you really want to play, build a cheap box at home and hook into the pots. Use it to send the telemarketers to the monkeys and pick up a couple of sip phones and test out things like customized rings and such. Makes it nice to know exactly who is calling during dinner. I don't answer when it says "Mother-In-Law Calling" |
| 15:20.21 | PodMan99a | i have a box at home this is what I am playing with... i just need to know how to register with a queue.... |
| 15:20.44 | PodMan99a | have x-lite... far from the best but its good to play with |
| 15:22.02 | [TK]D-Fender | PodMan99a: Keep reading, and pay attention to your dialplan, you've made a couple or really silly mistakes in there. |
| 15:22.06 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 15:22.24 | PodMan99a | is it the inbound number bit ..? |
| 15:24.53 | nestAr | PodMan99a: Registering with a queue, there's a couple different ways. I don't use pre-defined agents, I use AddQueueMember in my dialplan |
| 15:28.11 | PodMan99a | not as easy as that... any url's ?? |
| 15:29.05 | coppice | [TK]D-Fender: Pay attention to your dialplan? When your dialplan starts talking to you, take a rest |
| 15:29.33 | PodMan99a | lol |
| 15:29.36 | [TK]D-Fender | coppice: You're just jealous because the voices only talk to ME! :p |
| 15:30.05 | PodMan99a | [TK]D-Fender : they talk to me... but falling pixies are not very useful when programing asterix |
| 15:30.23 | [TK]D-Fender | PodMan99a: Ask for some of their magic dust. |
| 15:30.41 | PodMan99a | [TK]D-Fender: apparently is good to sniff |
| 15:30.45 | coppice | asterisk only works when you apply pixie dust |
| 15:33.04 | *** join/#asterisk Cyon (n=cyon@216.179.31.170) |
| 15:33.26 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 15:33.41 | iratik | Alright anyone familiar with freepbx? |
| 15:34.00 | PodMan99a | [TK]D-Fender: what url's would be good for this ... or what keywords would i enter in google... i have tried "asterisk dialplan config" |
| 15:34.03 | Darthclue | iratik, please go to #freepbx for help with that |
| 15:34.15 | iratik | sigh...... i'm not really sure if its a freepbx problem |
| 15:35.37 | nestAr | PodMan99a: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AddQueueMember |
| 15:35.38 | PBXX | is here someone ho now something about hylafax ? |
| 15:35.57 | coppice | nope. nobody ever used hylafax |
| 15:36.06 | Mw3 | i know it is a faxing software |
| 15:37.30 | Telemac | I've "[settings]\ sippeers => odbc.ast_cnf.sip_conf" in my extconfig.conf, so what could prevent "realtime load sipusers name _user" from finding matching row within Asterisk console ? |
| 15:37.35 | Darthclue | wasn't hylafax the boss on level 4 of asterisk the game? |
| 15:37.46 | _x86_ | anyone have any recommendations for CSU's? |
| 15:38.11 | _x86_ | need a CSU for a voice (CAS) T1, need rj48 both in from the smart jack and out to the PBX |
| 15:39.48 | [TK]D-Fender | Darthclue: In Soviet Russia, fax sends YOU. |
| 15:39.54 | nestAr | lol |
| 15:40.26 | Darthclue | [TK]D-Fender, if it sends me someplace warm, sunny, and quiet then I'm all for it. |
| 15:41.10 | asdx | [TK]D-Fender: lol slashdot culture :p |
| 15:41.35 | *** join/#asterisk IgorG (i=FeedomPa@host-90-188-188-119.pppoe.omsknet.ru) |
| 15:42.09 | [TK]D-Fender | iratik: And what would that be? |
| 15:43.37 | iratik | well my provider gave me the following settings for "extensions.conf" but when i look at extensions conf ... i'm not so sure where to put the settings |
| 15:43.39 | nestAr | i am so bored so early. |
| 15:43.48 | iratik | exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@icall) |
| 15:43.49 | nestAr | already replaced my furnace filter |
| 15:43.50 | iratik | [icall_in] |
| 15:43.51 | iratik | exten => 4175534249,1,Answer |
| 15:43.58 | iratik | and i don't know where to put that stuff |
| 15:44.50 | [TK]D-Fender | iratik: Yeah, thats dialplan BS and FreePBX's problem. |
| 15:44.53 | [TK]D-Fender | ~wglwat |
| 15:44.54 | jbot | it has been said that wglwat is well, good luck with all that |
| 15:46.33 | jameswf | ~rtfm |
| 15:46.34 | jbot | from memory, rtfm is Read The F*cking Manual (TM). It is a suggestion to do your homework before posting a question. Sometimes used as RTFM $SPECIFIC_MANUAL to refer to a specific source of information. See also http://en.wikipedia.org/wiki/RTFM |
| 15:46.36 | Telemac | I really need some clue about extconfig.conf and odbc. I'm certainly doing something bad but I really don't know what and where ... |
| 15:47.58 | Telemac | (and I've rtfm :) ) |
| 15:48.18 | *** join/#asterisk dijungal (n=kdaniel@63.175.159.171) |
| 15:48.43 | [TK]D-Fender | Telemac: then try this : |
| 15:48.46 | [TK]D-Fender | ~osmosis |
| 15:48.47 | jbot | i guess osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 15:49.19 | Darthclue | ~book |
| 15:49.20 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 15:50.17 | *** join/#asterisk pepo-- (n=pepOSX@190.78.220.149) |
| 15:50.31 | PodMan99a | [TK]D-Fender resolved ish... lol |
| 15:50.46 | [TK]D-Fender | PodMan99a: "ish".... slick..... |
| 15:51.16 | puzzled | tzafrir, have you also packaged oslec in your debs? |
| 15:51.25 | PodMan99a | i have to add the agents from the CLI |
| 15:51.31 | iratik | Whenever I make a call using any softphone i'm getting "Service Unavailable 503" ... what this might be a problem with? |
| 15:52.31 | _x86_ | http://cgi.ebay.com/AdTran-T1-CSU-ACE-3G-1203022L1-AC-Power-tested-warranty_W0QQitemZ170167735456QQihZ007QQcategoryZ80226QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 15:52.38 | _x86_ | will this work for a voice CAS T1? |
| 15:52.52 | *** join/#asterisk c0rnflake (n=c0rnflak@38.112.4.210) |
| 15:54.12 | [TK]D-Fender | _x86_: What do you want to do with your T1 exactly? |
| 15:54.19 | [TK]D-Fender | _x86_: And your link is dead |
| 15:56.05 | _x86_ | http://cgi.ebay.com/AdTran-T1-CSU-ACE-3G-1203022L1-AC-Power-tested-warranty_W0QQitemZ170167735456QQihZ007QQcategoryZ80226QQssPageNameZWDVWQQrdZ1QQcmdZViewItem |
| 15:56.13 | _x86_ | works for me... just used it |
| 15:56.48 | *** join/#asterisk russellb (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:56.48 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:57.09 | _x86_ | [TK]D-Fender: i want a test point between the smart jack and the asterisk PBX |
| 15:57.13 | [TK]D-Fender | _x86_: Didn't for me, but does now. |
| 15:57.31 | tzanger | http://spritesmods.com/?art=wcterror&f=had |
| 15:57.33 | tzanger | hahahahha |
| 15:57.37 | [TK]D-Fender | _x86_: What for? |
| 15:57.44 | _x86_ | don't want to do anything crazy like split channels off, etc... |
| 15:57.54 | _x86_ | [TK]D-Fender: by law in the US it's required |
| 15:58.02 | _x86_ | to have a CSU before your DTE |
| 15:58.30 | _x86_ | you're not allowed to run from smart jack to DTE directly, you need a CSU in there |
| 15:58.50 | coppice | by law? that's a strange thing to legislate. I can just imagine senators debating the merits of that :-) |
| 15:58.53 | [TK]D-Fender | _x86_: Ummmm..... CSU is integrated into Zaptel digital cards... |
| 15:58.58 | _x86_ | i had one from the 80's and it just died, so i need to replace it |
| 15:59.06 | _x86_ | [TK]D-Fender: perhaps digium cards, sure... i use sangoma |
| 15:59.22 | coppice | pretty much any modern card integrates the CSU |
| 15:59.28 | [TK]D-Fender | _x86_: Same bloody thing... uses the same Xilinx chip even IIRC. |
| 15:59.48 | coppice | the CSU is in the framer, not the Xilink |
| 16:00.05 | [TK]D-Fender | _x86_: And the claim of need for equipement before either of these is ridiculous. |
| 16:00.12 | [TK]D-Fender | coppice: Noted :) |
| 16:00.41 | *** join/#asterisk pepo--- (n=pepOSX@190.72.153.45) |
| 16:06.14 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 16:07.35 | destructure | anybody done any testing on the accuracy of asterisk timers? if I wait(30) how close to 30 will it be? particularly under load |
| 16:07.40 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 16:08.56 | *** join/#asterisk dhill (n=dhill@fog.mindcry.org) |
| 16:09.18 | ai-a | destructure: you could do it. |
| 16:09.29 | dhill | is there a way to edit asterisk voicemail? For example, I want to get rid of the option to set the unavailable message.. |
| 16:09.53 | ai-a | dhill: i would like to completly edit the whole callflow of voicemail.. |
| 16:10.05 | dhill | yea |
| 16:10.10 | destructure | ai-a: thanks for the encouragement! |
| 16:10.21 | ai-a | go destructure.. go destructure. |
| 16:10.51 | [TK]D-Fender | dhill: You have the source... get coding. |
| 16:10.54 | dhill | I use asterisk for customers not inner-office. Customers do not understand unavailable and busy. I'd rather just have one.. make it easy... |
| 16:11.17 | ai-a | Fender: is the voicemail possible to reproduce in an ivr ? |
| 16:11.47 | [TK]D-Fender | ai-a: Sure, been done before. |
| 16:12.20 | ai-a | or, better still more control over the vm from within the config file ;) |
| 16:12.31 | [TK]D-Fender | ai-a: Sure... see my answer for dhill above :p |
| 16:12.36 | ai-a | Heh :p. |
| 16:13.28 | *** join/#asterisk muiro (n=dan@host-69-48-104-2.akr.choiceone.net) |
| 16:14.44 | muiro | question: What I'm trying to do is allow currency entry. I want to match any number of numbers, then a star, then two numbers the pound |
| 16:14.56 | *** join/#asterisk alvariux (n=alvaro@189.158.163.157) |
| 16:15.03 | alvariux | hello |
| 16:15.11 | muiro | _X.*XX# however isn't working |
| 16:15.21 | [TK]D-Fender | muiro: All doable in an IVR. |
| 16:15.27 | muiro | it seems like asterisk doesn't want to match the * or the # |
| 16:15.34 | [TK]D-Fender | muiro: And NO, you can't give a pattern like that. |
| 16:15.57 | [TK]D-Fender | muiro: you'll have to collect each digit and parse the heck out of it. |
| 16:16.06 | [TK]D-Fender | muiro: Not even that complicated actually... |
| 16:16.09 | coppice | he can do anything he wants |
| 16:16.16 | [TK]D-Fender | muiro: More like parse a bit ;) |
| 16:16.17 | ai-a | or write agi script :) |
| 16:16.38 | muiro | yeah, I was really hoping just to do it with pattern matching but I guess parsing it is |
| 16:16.47 | muiro | I got the reading out of currency amounts working smoothly, btw |
| 16:16.56 | muiro | it's just the entry that's a bit loopy |
| 16:17.03 | alvariux | hi, somebody uses phpagi? |
| 16:17.04 | [TK]D-Fender | muiro: only about 5 lines of dialplan :) |
| 16:17.10 | muiro | exactly |
| 16:17.39 | muiro | if it weren't for users I could leave it alone as it stands, but that wouldn't work |
| 16:17.56 | puzzled | anyone have a recommendation which function I could use best with 1.4 to see if the sipheader(to) contains a fqdn or ip address? |
| 16:18.17 | muiro | I mean, the prompt says input it this way, that's the way you do it. Users won't, though. |
| 16:19.14 | muiro | they're like drugs, really. Users. |
| 16:19.20 | muiro | You need them but you hate them. |
| 16:19.26 | [TK]D-Fender | muiro: What I'd advise is to tell them to entire the entire amount WIOUT a decmil indicator and dvide out the last 2 digits. (/100) |
| 16:19.40 | [TK]D-Fender | muiro: dead easy that way and a no-brainer. |
| 16:19.45 | muiro | yeah |
| 16:20.13 | muiro | like an atm would do it |
| 16:20.19 | [TK]D-Fender | muiro: Sure. |
| 16:20.30 | [TK]D-Fender | muiro: why complicate things for morons? Then jsut read it back to confirm. |
| 16:20.35 | muiro | I hesitated to do that initially because it may be confusing without that visual decimal mark |
| 16:20.51 | [TK]D-Fender | muiro: VISUAL?! We're talking about a phone IVR here. |
| 16:20.52 | muiro | or, at least, me being a visual thinker, I thought it might be confusing |
| 16:21.09 | muiro | yes, that's why I avoided that, because there was no visual thing |
| 16:21.14 | muiro | "like an atm", as it were |
| 16:22.46 | muiro | I had remembered dialing into an ivr to make a tax payment for a company I was working for one time and I remember entering the decimal as *, I liked that. I was hoping to "force" it using the matching but I guess I'll just parse it. |
| 16:22.57 | muiro | no big deal |
| 16:23.20 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 16:24.35 | [TK]D-Fender | muiro: you can, but you'd have to code about 1-2 dozen lines of IVR to supoprt this. |
| 16:24.37 | muiro | [TK]D-Fender: thanks for the advice |
| 16:24.55 | [TK]D-Fender | Darthclue: load res.psychic.so :p |
| 16:25.18 | Darthclue | I tried. It kept saying, failure to connect, no signal. |
| 16:25.34 | muiro | more like no carrier... |
| 16:27.01 | coppice | try www.spiritdsp.com |
| 16:30.18 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:30.30 | *** join/#asterisk Kandinsky (n=Kandinsk@perla2.tm.ew.ro) |
| 16:30.47 | _x86_ | [TK]D-Fender: did you say anything on my CSU? |
| 16:31.08 | [TK]D-Fender | _x86_: Yeah I said you don't need one :) |
| 16:31.22 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 16:31.27 | puzzled | does voicemail also require ztdummy? I always forget when ztdummy is needed |
| 16:31.45 | *** join/#asterisk dswillia (n=me2@199.3.247.34) |
| 16:31.46 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 16:31.50 | Mw3 | puzzled: meetme, iax trunking |
| 16:31.58 | puzzled | ta |
| 16:32.14 | deeperror | is there a dialplan variable that tells me the sip username that is making a call? |
| 16:32.25 | *** join/#asterisk JayTee52 (n=jforde05@207-67-84-181.static.twtelecom.net) |
| 16:32.44 | JayTee52 | hello all |
| 16:32.58 | [TK]D-Fender | deeperror: if it matched a sip.conf entry it'd be in the channel name... |
| 16:33.33 | dswillia | hey all is there a way to tell asterisk to compile with zaptel and libpri support. I have d/led the latest zap, libpri, and Asterisk. Compiled both lib, then zap. When I compile asterisk "make install" it works no errors, but when i connect to the cli I have no options for zap or pri |
| 16:33.37 | puzzled | how about ${SIP_HEADER(From)} |
| 16:34.02 | puzzled | dswillia, first compile & install zaptel, then libpri, then asterisk |
| 16:34.31 | dswillia | what is the best way to "uninstall" asterisk? |
| 16:34.57 | puzzled | dunno. did you try # make uninstall ? |
| 16:35.27 | dswillia | wow thank you |
| 16:35.36 | dswillia | that was painless |
| 16:35.45 | puzzled | lucky guess :) |
| 16:36.35 | muiro | also dswillia make sure zap is running |
| 16:37.58 | deeperror | puzzled: that did it for me thanks....fender: the channel variable didn't seem to work for my application not sure what the difference was |
| 16:38.32 | [TK]D-Fender | deeperror: clarify "didn't work" |
| 16:38.32 | Kandinsky | in zapata.conf, can i use multiple contexts if i want to use all the channels in every context? |
| 16:38.52 | [TK]D-Fender | Kandinsky: that makes no sense. |
| 16:39.04 | Kandinsky | why? |
| 16:39.29 | Kandinsky | i want to have both incoming and outgoing calls on all the channels |
| 16:39.36 | [TK]D-Fender | Kandinsky: You do not use "zaptel channels" in DIALPLAN contexts. In the dialplan you USE zaptel channels, either by fixed number, or by a GROUP taht you make them a member of. |
| 16:39.49 | deeperror | fender: was using it for a name in mixmonitor and the recording would not save probably the slash or something? |
| 16:40.20 | [TK]D-Fender | Kandinsky: the context you set in zapata is where * will send INCOMING calls from that channel. This can only be a SINGLE place. |
| 16:40.34 | Kandinsky | aha |
| 16:40.39 | [TK]D-Fender | deeperror: "show function CUT" <---- |
| 16:41.01 | Kandinsky | so all my channels can be only in the incoming context |
| 16:41.20 | Kandinsky | and when i want to dial out..i include the incoming context? |
| 16:41.21 | [TK]D-Fender | Kandinsky: you are not getting it. |
| 16:41.26 | Kandinsky | nope :( |
| 16:41.32 | Darthclue | ~book |
| 16:41.33 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 16:41.35 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 16:41.38 | [TK]D-Fender | Kandinsky: this ha NOTHING to do with OUTGOING CALLS. |
| 16:41.58 | Kandinsky | but what are those channels in zapat.conf actually? |
| 16:42.02 | Kandinsky | not my isdn channels? |
| 16:42.11 | Kandinsky | which i can use |
| 16:42.22 | [TK]D-Fender | Kandinsky: You can have 1,000,000 contexts in extensions.cfon that each have extens that DIAL out your zaptel channels, but that has nothing to do with how * handles INCOMING calls |
| 16:42.33 | Kandinsky | i know that |
| 16:42.47 | Kandinsky | i am speaking strictly on the [] in the zapata.conf |
| 16:43.13 | [TK]D-Fender | Kandinsky: there are no [] <- (context looking things) in zapata.conf |
| 16:43.21 | dswillia | how would i check to see if zap is running? |
| 16:43.25 | deeperror | fender: thanks for CUT |
| 16:43.25 | [TK]D-Fender | Kandinsky: that isn't how channels are defined. |
| 16:43.30 | Kandinsky | context=incoming |
| 16:43.30 | Kandinsky | channel => 1-2 |
| 16:43.30 | Kandinsky | channel => 4-5 |
| 16:43.34 | Kandinsky | so this is wrong? |
| 16:43.43 | Kandinsky | if i have this in zapata.conf |
| 16:45.00 | Kandinsky | do you happen to know any good documentation for zapata.conf when using BRI isdn with hfc-s chipsets? |
| 16:45.13 | Kandinsky | i understand the contexts in asterisk |
| 16:45.23 | Kandinsky | but not sure about the zapata thing |
| 16:45.32 | Kandinsky | and the link between the 2 |
| 16:46.02 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 16:46.02 | *** mode/#asterisk [+o anthm] by ChanServ |
| 16:46.33 | Kandinsky | [TK]D-Fender: u still there? |
| 16:46.48 | *** part/#asterisk harpal (n=Harpal@124.125.255.223) |
| 16:47.00 | [TK]D-Fender | Kandinsky: pastebin your zapata |
| 16:47.02 | [TK]D-Fender | ~pb |
| 16:47.02 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 16:49.33 | *** join/#asterisk bantu (n=Miranda@p54A32965.dip0.t-ipconnect.de) |
| 16:50.00 | Kandinsky | http://pastebin.com/d57674b4c |
| 16:50.35 | Kandinsky | i left some comments from my previous atempts |
| 16:50.58 | Kandinsky | so ..i have 4 isdn channels |
| 16:51.00 | [TK]D-Fender | You have 4 channels defined and no groups. All incoming calls on those channels go to the same context |
| 16:51.09 | Kandinsky | yes |
| 16:51.18 | Kandinsky | tell me more about these groups pls |
| 16:51.51 | Kandinsky | what they are good at and how to use them |
| 16:52.16 | Kandinsky | what i want..is to have all the 4 channels available for calling or being called |
| 16:53.14 | [TK]D-Fender | Kandinsky: You can group your channels together so that you can have * automatically choose a free channel to call out of when placing an outgoing call |
| 16:53.40 | Kandinsky | so i group all my channels in group 1 for instance |
| 16:54.05 | Kandinsky | because that coresponds to what i want |
| 16:54.06 | Kandinsky | no? |
| 16:54.51 | [TK]D-Fender | Kandinsky: Currently they are NOT grouped. |
| 16:54.57 | Kandinsky | i know :) |
| 16:55.06 | Kandinsky | i mean that is what i must do |
| 16:55.14 | [TK]D-Fender | Kandinsky: But that is what I WOULD do typically if all of your channels can be treated as equal. |
| 16:55.35 | Kandinsky | ok..solved that |
| 16:55.54 | Kandinsky | and what is the order? |
| 16:55.56 | *** join/#asterisk docelmo (n=vircuser@c-76-99-153-242.hsd1.de.comcast.net) |
| 16:55.57 | Kandinsky | if it matters |
| 16:56.04 | Kandinsky | group ... context..channels? |
| 16:58.13 | [TK]D-Fender | Kandinsky: if you give them all the same group membership you can dial out the group like Dial(Zap/g1/1234567890) |
| 16:58.31 | [TK]D-Fender | Kandinsky: this has nothing to do with CONTEXT, we are only talking about GROUP right now |
| 16:58.31 | Kandinsky | yeah..i know that |
| 16:58.48 | Kandinsky | but i am talknig abouty the syntax in zapata.conf |
| 16:59.01 | Kandinsky | i first must have a group |
| 16:59.04 | Kandinsky | then a context |
| 16:59.11 | Kandinsky | and then the channels defined? |
| 16:59.44 | [TK]D-Fender | Kandinsky: you already have a context set and you'll probably want to handle all incoming calls the same way, so thats fine. what you need to do is simply set the GROUP |
| 16:59.55 | Kandinsky | ok |
| 17:00.09 | Kandinsky | i was talking about a general case |
| 17:00.15 | Kandinsky | i just set my group |
| 17:00.30 | [TK]D-Fender | Kandinsky: what is "a general case"? |
| 17:00.42 | Kandinsky | aaa.....common case |
| 17:00.50 | Kandinsky | never mind |
| 17:01.03 | *** join/#asterisk irule (n=irule@200.53.61.4) |
| 17:01.06 | Kandinsky | i want to understand the logic in zapata.conf |
| 17:01.22 | Kandinsky | so.. in group 1 i have all my 4 channels |
| 17:01.41 | Kandinsky | and when incoming calls come..they will go to my incoming context in extensions.conf |
| 17:01.43 | Kandinsky | right? |
| 17:01.46 | *** join/#asterisk rpm (n=russell@75.153.47.179) |
| 17:01.52 | [TK]D-Fender | Kandinsky: Correct |
| 17:01.57 | Kandinsky | :) |
| 17:02.08 | Kandinsky | ok...done with zapata.conf? |
| 17:02.18 | Kandinsky | anything else i should consider adding here? |
| 17:02.20 | [TK]D-Fender | Kandinsky: and to pick any free channel to dial OUT of you'd use a Dial command like I showed you earlier |
| 17:02.27 | Kandinsky | yes |
| 17:02.33 | Kandinsky | now |
| 17:02.43 | [TK]D-Fender | Kandinsky: Nope, thats it if this is how you want to handle your calls. |
| 17:03.04 | Kandinsky | ok...other stuff now :) |
| 17:03.08 | Kandinsky | extensions.conf |
| 17:03.31 | Kandinsky | in the incoming context |
| 17:03.47 | Kandinsky | i must defin where all my calls will ring |
| 17:03.52 | Kandinsky | correct? |
| 17:04.17 | Kandinsky | i am having a major problem with my incoming context |
| 17:04.40 | Kandinsky | i want to define 10 specific numbers (our phone office numbers) |
| 17:05.02 | Kandinsky | so that each user will have its own number |
| 17:05.19 | Kandinsky | so when someone is calling xxxx22 |
| 17:05.24 | Kandinsky | it will ring user1 |
| 17:05.27 | Kandinsky | so when someone is calling xxxx23 |
| 17:05.29 | Kandinsky | it will ring user2 |
| 17:05.30 | Kandinsky | etc |
| 17:05.50 | Kandinsky | until xxxx32 for example |
| 17:06.01 | Kandinsky | now...my problem |
| 17:06.28 | Kandinsky | if i set exten => _X!22,1,Dial(SIP/user1,20) |
| 17:06.39 | Kandinsky | in the incoming context I am fine |
| 17:06.48 | Kandinsky | but i want to be more specific |
| 17:06.55 | Kandinsky | to define the whole number actually |
| 17:07.12 | Kandinsky | for instance that xxxx22 is 111122 |
| 17:07.17 | Kandinsky | local number |
| 17:07.33 | Kandinsky | but if I type that..the phone dosen't ring |
| 17:07.53 | Kandinsky | and I don't know how asterisk actually receives the number dialed from the outside |
| 17:08.20 | Kandinsky | i tried to find out the number of digits * reads for 111122 |
| 17:09.14 | Kandinsky | by setting _XXXXXX |
| 17:09.24 | Kandinsky | and it wouldn't ring |
| 17:09.30 | Kandinsky | and i kept on going |
| 17:09.36 | Kandinsky | increasing the X number |
| 17:09.43 | Kandinsky | but no pattern would be fine |
| 17:10.03 | Kandinsky | any ideas how i can find the number asterisk is really trying to dial in my context? |
| 17:10.09 | deeperror | I call 2 inbound did numbers from the same provider. They both goto the same locations in my dialplan and one doesn't jump to options in the ivr the other one does. I can see dtmf being identified in messages but it doesn't seem to make any actions. This is running over iax protocol. Any suggestions? |
| 17:13.30 | deeperror | here is what I see in msg's http://pastebin.ca/775538 |
| 17:14.46 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 17:16.30 | [TK]D-Fender | Kandinsky: You'll have to paste your dialplan and the CLI output of a failed call at verbose 10 so we can debug. |
| 17:17.09 | Kandinsky | ok...thanks ...but i have to do something now...will be back in 15 min i guess |
| 17:17.40 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 17:17.40 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 17:19.18 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 17:22.10 | *** join/#asterisk davidcsi (n=davidcsi@180.Red-213-97-249.staticIP.rima-tde.net) |
| 17:22.53 | davidcsi | hello all, question: I changed the listening port for the manager.conf.... how do i unload and load the manager for the port change to tale effect?? |
| 17:25.15 | davidcsi | anyone? |
| 17:25.30 | deeperror | asterisk restart? |
| 17:25.42 | davidcsi | doesn't work |
| 17:25.49 | davidcsi | reload doesn't work |
| 17:26.29 | deeperror | asterisk -rx "restart now" |
| 17:26.35 | *** join/#asterisk rhombus (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com) |
| 17:27.29 | rhombus | The Polycom phones can be told to reboot with a "sip notify check-cfg <registration>" on the Asterisk CLI |
| 17:27.41 | rhombus | is there a way to accomplish the same thing with the Aastra sets? |
| 17:27.47 | rhombus | (specifically, the 480i) |
| 17:29.15 | davidcsi | deeperror: does that drops the calls? |
| 17:29.37 | deeperror | yes it will |
| 17:29.45 | deeperror | asterisk -rx "restart when convenient" |
| 17:29.54 | davidcsi | then I can't do that |
| 17:30.02 | davidcsi | is there any other way? |
| 17:30.05 | deeperror | no |
| 17:30.19 | deeperror | has to restart asterisk for that type of change |
| 17:30.21 | *** join/#asterisk bantu (n=Miranda@p54A32965.dip0.t-ipconnect.de) |
| 17:30.38 | davidcsi | do you know which module handles the manager? |
| 17:30.55 | deeperror | that i'm not sure |
| 17:31.07 | deeperror | http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20manager.conf |
| 17:31.21 | deeperror | "Simply reloading asterisk will not enable the manager. You must shut down asterisk and restart." |
| 17:31.28 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 17:31.34 | Assid | hey |
| 17:31.42 | Assid | anyone here have 2 fwd accounts on the same box |
| 17:33.39 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 17:33.52 | De_Mon | queues have members, what do you call the people on hold? |
| 17:34.00 | De_Mon | queue callers sounds gay |
| 17:34.15 | coppice | suckers |
| 17:34.25 | deeperror | HAHA |
| 17:34.31 | De_Mon | yeah... thats not much better |
| 17:34.43 | deeperror | member holders |
| 17:34.57 | coppice | lusers |
| 17:35.10 | [TK]D-Fender | deeperror: lol |
| 17:35.12 | coppice | friends of the telco |
| 17:35.24 | [TK]D-Fender | deeperror: now THAT was gay |
| 17:35.26 | De_Mon | queue waiters |
| 17:36.03 | coppice | sheep |
| 17:36.10 | deeperror | haha |
| 17:36.24 | De_Mon | all these silly answers tells me nobody has a serious answer |
| 17:36.32 | deeperror | unless gender is taken into account of the member holder status |
| 17:36.50 | [TK]D-Fender | De_Mon: Measn nobody bothered to invent a term for that specific scenario. |
| 17:36.53 | De_Mon | members are the people the queue is calling |
| 17:36.55 | Qwell | there we go - gender based queue priorities |
| 17:36.57 | coppice | i thought lusers and suckers were the serious answers |
| 17:37.02 | davidcsi | deeperror: ok, thanks |
| 17:37.06 | Assid | freaking odd. i registered for a new fwd account.. and that just refuses to register |
| 17:37.31 | [TK]D-Fender | Assid: And you're doing SO much to help yourself here too.... a shame... |
| 17:37.32 | deeperror | i'm hearing a lot about fwd accounts...does this give me a number? or is it similar to grand central? |
| 17:37.57 | Assid | [TK]D-Fender hehe.. okay would like some help.. 1 account works. the other one doesnt.. |
| 17:38.09 | De_Mon | well, luser is slang, and sucker is just, bad. You're fired! |
| 17:38.14 | Assid | [TK]D-Fender and yes i am using the right password on them |
| 17:38.24 | De_Mon | don't even bother asking about that last paycheck... |
| 17:39.01 | De_Mon | I thought for sure the telephony world had these terms defined hundreds of years ago |
| 17:43.50 | *** part/#asterisk davidcsi (n=davidcsi@180.Red-213-97-249.staticIP.rima-tde.net) |
| 17:45.13 | rhombus | De_Mon: Just use callers and get over your silly discomfort about it |
| 17:45.27 | rhombus | De_Mon: it says what it means; not everything has to be jargoned up |
| 17:45.58 | rhombus | callers, agents -- end of discussion |
| 17:46.12 | rhombus | Now, my Aastra phones. Can I reboot them from the Asterisk CLI? |
| 17:48.00 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:48.02 | *** join/#asterisk _ys (n=yuri@236-069.nat.mns.ru) |
| 17:50.01 | [TK]D-Fender | rhombus: look harder next time : http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip_notify.conf |
| 17:50.47 | rhombus | Umpteen google searches didn't turn that up |
| 17:51.20 | rhombus | how much harder am I supposed to look? how much time am I supposed to waste? Now I know, and the next time somebody asks, I can answer also |
| 17:51.24 | rhombus | nm |
| 17:51.40 | [TK]D-Fender | ~wikis |
| 17:51.41 | jbot | from memory, wikis is http://www.voip-info.org |
| 17:51.45 | [TK]D-Fender | ^^^^^^^^^^ |
| 17:51.56 | rhombus | [TK]D-Fender: thanks for being patronizing |
| 17:52.02 | *** join/#asterisk Seldon75 (n=chatzill@69.77.161.3) |
| 17:52.12 | rhombus | [TK]D-Fender: I've been there... MANY times :) |
| 17:52.24 | rhombus | try this |
| 17:52.30 | [TK]D-Fender | rhombus: rhombus And http://www.google.ca/search?hl=en&q=aastra+SIP+remote+reboot&btnG=Search&meta= |
| 17:52.31 | rhombus | do a search, on google, for |
| 17:52.40 | rhombus | "reboot Aastra from Asterisk CLI" |
| 17:52.41 | [TK]D-Fender | rhombus: was #5. Like I said.... look harder |
| 17:52.47 | rhombus | and see what turns up |
| 17:53.02 | Seldon75 | Hi, I want to write a cron-job that does a soft-hangup on any lines that are up for longer than 2 hours. |
| 17:53.12 | Seldon75 | so how do I ask asterisk for a Zap channel's up-time? |
| 17:53.57 | Seldon75 | ...without going into the console |
| 17:54.01 | [TK]D-Fender | Seldon75: "show channel [channel]" |
| 17:54.04 | Qwell | telepathy |
| 17:54.20 | rhombus | <PROTECTED> |
| 17:54.25 | rhombus | that's a useless piece of advice |
| 17:54.26 | rhombus | thanks |
| 17:54.43 | *** join/#asterisk steliosk (n=Stelios@ipa226.211.tellas.gr) |
| 17:54.48 | [TK]D-Fender | rhombus: I wanted to make sure not to read how to reboot their ANALOG phones :p |
| 17:54.57 | [TK]D-Fender | rhombus: Your Google-fu is weak... |
| 17:54.57 | deeperror | seldon75: why not just set a timeout for 2h then not worry about it |
| 17:55.10 | *** join/#asterisk kmchen (n=kmchen@gar13-4-82-240-99-84.fbx.proxad.net) |
| 17:55.21 | Seldon75 | deeperror: sounds good - would that be in zapata.conf? |
| 17:55.49 | rhombus | I think that "reboot Aastra from Asterisk CLI" is an adequate distinction that will protect me from instructions on how to reboot the ADSI sets |
| 17:56.02 | ai-a[wrk] | rhombus: why do you think its asterisk that performs this ? Maybe its your telephone company that can reboot your phone ? |
| 17:56.03 | rhombus | anyway |
| 17:56.12 | [TK]D-Fender | rhombus: thinking : yet another skill best left to trained professionals ;) |
| 17:56.26 | deeperror | seldon: Set(TIMEOUT(absolute)=1800) in dialplan |
| 17:56.27 | rhombus | [TK]D-Fender: are you always this arrogant? |
| 17:56.28 | [TK]D-Fender | rhombus: Oh... and you're welcome :) |
| 17:56.33 | ai-a[wrk] | Heh. |
| 17:56.34 | Assid | err is there a speach to text module in asterisk.. that lets you convert the voicemail into text and email it |
| 17:56.43 | Seldon75 | how can I set a Zap timeout? |
| 17:56.52 | rhombus | It's easier to be thankful to people with some largesse themselves |
| 17:56.59 | Seldon75 | ok |
| 17:57.10 | [TK]D-Fender | rhombus: Arrogant? I was thinking more like witty, sarcastic, and jovial. |
| 17:57.27 | Seldon75 | deeperror: will that work for i/c calls? |
| 17:57.28 | coppice | Assid: you need app_magic_pixies for that |
| 17:57.37 | Assid | haha |
| 17:57.48 | deeperror | that i'm not sure |
| 17:57.50 | [TK]D-Fender | rhombus: I included smilies, pop-culture references and the solution! |
| 17:57.51 | kmchen | Hi evrybody. Could someone help me to configure musiconhold for streaming radio ? |
| 17:57.51 | Assid | i keep telling this guy no such thing exists.. hes like no.. it does |
| 17:58.03 | deeperror | i use it when zap lines are making outbound calls thru sip/iax |
| 17:58.34 | Seldon75 | deeperror: yeah I need it only to work for I/C calls... |
| 17:58.36 | [TK]D-Fender | rhombus: All part of the service, please claim your KY discount coupon before you log out! ;) |
| 17:58.39 | rhombus | [TK]D-Fender: your memory is selective -- have a closer look at what you typed |
| 17:58.48 | kmchen | I configured musiconhold.conf with dir=/var/lib/asterisk/mohmp3-empty |
| 17:58.48 | kmchen | application=mpg123 -q -r 8000 -f 8192 -b 2048 -s --mono http://213.205.96.91:9915/ |
| 17:59.12 | kmchen | but total silence |
| 17:59.15 | deeperror | just place it in the dialplan before the calls you want to timeout |
| 17:59.30 | [TK]D-Fender | rhombus: Trust me you'd know if I was seriously chewing someone out. Heck the entire channel would. Anyways, happy coding :) |
| 17:59.33 | kmchen | I did that |
| 18:00.43 | Seldon75 | deeperror: so, like "exten => s,1,Set(TIMEOUT(absolute)=1800) " and then "exten => s,2,Goto(incoming,s,1)"... |
| 18:00.44 | [TK]D-Fender | Seldon75: just cron up a job to dump your channels and kill off the old ones. |
| 18:01.12 | Seldon75 | [TK]D-Fender: so you would recommend the timeout option deeperror is suggesting? why not? |
| 18:01.31 | kmchen | deeperror: exten => 1,1,SetMusicOnHold(native) |
| 18:01.31 | kmchen | ;exten => t,1,WaitMusicOnHold(3) |
| 18:01.31 | kmchen | exten => t,2,Dial(SIP/keynux) |
| 18:02.17 | *** join/#asterisk muiro (n=muiro@cpe-76-189-84-108.neo.res.rr.com) |
| 18:02.40 | [TK]D-Fender | Seldon75: Should work, but you'd have to add it to every originating exten. |
| 18:03.18 | Kandinsky | back |
| 18:03.28 | Seldon75 | [TK]D-Fender: you mean to prevent the timeout from 'hanging around' once set for a particular call type |
| 18:03.38 | Kandinsky | [TK]D-Fender: another thing i forgot |
| 18:03.43 | Kandinsky | about zapata.conf |
| 18:03.50 | [TK]D-Fender | Seldon75: meaning you've have to insert that all over the place. |
| 18:03.53 | Seldon75 | :q |
| 18:03.59 | Kandinsky | prilocaldialplan |
| 18:04.05 | Kandinsky | does it set anything? |
| 18:04.19 | [TK]D-Fender | Kandinsky: Not sure on this. I don't do BRI |
| 18:04.30 | Kandinsky | ok |
| 18:04.48 | Kandinsky | but do you know about international prefix and nationalprefix? |
| 18:04.50 | Seldon75 | ok, so how can I execute a console command and get the output from within a cron-job? |
| 18:05.30 | deeperror | asterisk -rx "your command" |
| 18:05.34 | Seldon75 | thx |
| 18:06.03 | Kobaz | is there any other info besids iax2 show registry to check the status of iax trunks |
| 18:07.03 | Seldon75 | :qls |
| 18:09.36 | [TK]D-Fender | Kobaz: "iax2 show peers" |
| 18:10.58 | Kobaz | yeah and besids that one too |
| 18:11.05 | Kobaz | everything checks out find |
| 18:11.06 | Kobaz | fine |
| 18:11.20 | Kobaz | but this voicepulse number isn't hitting asterisk |
| 18:11.21 | [TK]D-Fender | Kobaz: What else could there possibly be? |
| 18:11.25 | Kobaz | no idea |
| 18:11.38 | Kobaz | iax soft phones can dial other iax trunks |
| 18:11.45 | [TK]D-Fender | Kobaz: got the proper ports forwarded to *? |
| 18:12.02 | Kobaz | the box is set up as a DMZ |
| 18:12.14 | [TK]D-Fender | Kobaz: And jsut because you're registered doesn't meant hat you haven't bungled up your DIALPLAN, etc.... |
| 18:12.27 | Kobaz | it's not related to dialplan |
| 18:12.30 | [TK]D-Fender | Kobaz: enable iax2 debug and pastebin the CLI output of a failed call attempt |
| 18:12.31 | Kobaz | it's not even getting that far |
| 18:12.35 | Kobaz | it's not accepting the call |
| 18:12.40 | Kobaz | it's not even getting the call |
| 18:12.53 | [TK]D-Fender | Kobaz: dialplan failure CAN prevent you from getting a call. |
| 18:12.59 | deeperror | VP numbers require a 1 in front of them as well on incoming |
| 18:13.03 | Kobaz | verbosity is 6 |
| 18:13.05 | [TK]D-Fender | Kobaz: do do as I suggested. |
| 18:13.14 | [TK]D-Fender | Kobaz: I said "iax2 debug", not "verbose" |
| 18:13.17 | Kobaz | nothing happens when the number is dialed |
| 18:13.19 | Kobaz | yeah |
| 18:13.37 | Kobaz | oh |
| 18:13.37 | Kobaz | hmm |
| 18:13.39 | Kobaz | i did get stuff |
| 18:13.39 | Kobaz | yay |
| 18:14.04 | Kobaz | okay this helps |
| 18:15.22 | [TK]D-Fender | Kobaz: You go quietly fix it now. I suspect you are feeling rather silly now.... |
| 18:16.45 | Kandinsky | [TK]D-Fender: can i see in asterisk wich outside number is trying to dial? |
| 18:17.00 | Kobaz | :( |
| 18:17.04 | Kandinsky | which |
| 18:17.05 | Kobaz | rather silly |
| 18:17.12 | [TK]D-Fender | Kandinsky: ummmm reword that please... taht didn't add up./ |
| 18:17.39 | Kandinsky | if someone on the outside is dialing a number |
| 18:17.44 | Kandinsky | a customer for instance |
| 18:17.55 | Kandinsky | and i want to see what number asterisk receives |
| 18:18.16 | deeperror | {$EXTEN} |
| 18:18.31 | Kandinsky | to see if asterisk receives local coda+number or just number ... |
| 18:18.36 | Kandinsky | if u get the ideea |
| 18:19.29 | [TK]D-Fender | Kandinsky: You get that number as the exten they have dialed. thats what you're pattern-matching for. |
| 18:19.44 | Kandinsky | if i set exten => 101055,1,Dial(SIP/user,20) |
| 18:20.00 | Kandinsky | and someone from the same telephone network dials 101055 |
| 18:20.06 | Kandinsky | without the local code |
| 18:20.06 | [TK]D-Fender | Kandinsky: in that example, you KNOW they dialed 101055 if that line is being executed |
| 18:20.21 | Kandinsky | but asterisk doesn't seem to know |
| 18:20.27 | [TK]D-Fender | Kandinsky: So make a bunch of extens to match what you want. |
| 18:20.29 | Kandinsky | because the phone isn't ringing |
| 18:20.51 | Kandinsky | it only works with something like _X!309055 for instance |
| 18:20.52 | [TK]D-Fender | Kandinsky: enable debug and see what # * is receiving. |
| 18:21.04 | Kandinsky | tips on how to do that pls |
| 18:21.12 | Kandinsky | sorry |
| 18:21.18 | Kandinsky | it only works with something like _X!101055 for instance |
| 18:21.19 | [TK]D-Fender | Kandinsky: And try things like using a more general pattern match as opposed to a hard-coded number like that |
| 18:21.33 | deeperror | isn't this in the manual? |
| 18:21.59 | Kandinsky | but i need hard coded numbers :) because i want to particularize |
| 18:22.12 | [TK]D-Fender | Kandinsky: please re-read chapter 5 a few more times so you can see what kind of pattern matching best suits your needs and go TEST what the telco sends you. |
| 18:22.14 | [TK]D-Fender | ~book |
| 18:22.21 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 18:22.22 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 18:22.38 | Kandinsky | ok..that's what i'm reading AGAIN now |
| 18:22.40 | *** join/#asterisk gabiru (n=gabiru@213.37.159.28.dyn.user.ono.com) |
| 18:22.46 | Kandinsky | pls tell me about that debuging stuff |
| 18:22.47 | [TK]D-Fender | Kandinsky: And realize you can have a call land on a more general place and the TEST for speciifc value to determine how to proceed |
| 18:23.14 | Kandinsky | to see how asterisk receives the number dialed |
| 18:23.21 | [TK]D-Fender | Kandinsky: not sure how it works for BRI, but for pri it would be "pri debug span [span]" |
| 18:23.54 | Kandinsky | aha |
| 18:23.54 | [TK]D-Fender | Kandinsky: but do a GENERAL pattern match like "_x." first and NoOp the ${EXTEN} to see what it looks like for a variety of calls. |
| 18:24.41 | Kandinsky | exten => _X.,1,.....??? |
| 18:25.08 | Kandinsky | can u please tell me exactley how that should look like? |
| 18:25.23 | [TK]D-Fender | Kandinsky: exten => _x.,1,NoOp(Incoming call from exten # "${EXTEN}") |
| 18:25.29 | Kandinsky | k |
| 18:25.29 | Kandinsky | 10x |
| 18:26.01 | Kandinsky | and what exactley does that NoOp do? |
| 18:26.01 | [TK]D-Fender | Kandinsky: Now stop and go back and reread the chapter about the dialplan, flow control, over and over and just TRY STUFF. |
| 18:26.14 | Kandinsky | can i see some output? |
| 18:26.29 | [TK]D-Fender | Kandinsky: It just prints a line in the * console so yuo can SEE something. It doesnt' actually "DO" anythiner |
| 18:26.30 | Kandinsky | or that whould show if i have debug turned on? |
| 18:26.33 | [TK]D-Fender | anything* |
| 18:26.34 | Kandinsky | a |
| 18:26.35 | Kandinsky | ok |
| 18:26.37 | Kandinsky | got it |
| 18:26.46 | [TK]D-Fender | Kandinsky: You'll see it iver you're at verbose 3 or higher IIRC |
| 18:26.58 | [TK]D-Fender | Kandinsky: Now go test, go read, then test some more. |
| 18:27.06 | Kandinsky | :) |
| 18:27.48 | *** join/#asterisk Dovid (n=Dovid@bzq-88-153-142-81.red.bezeqint.net) |
| 18:27.52 | *** part/#asterisk pepo--- (n=pepOSX@190.72.153.45) |
| 18:28.00 | Dovid | anyone here set up BLF on a snom 360 before / |
| 18:28.02 | Dovid | ?* |
| 18:39.40 | *** join/#asterisk bantu (n=Miranda@p54A316EC.dip0.t-ipconnect.de) |
| 18:51.10 | Kandinsky | [TK]D-Fender: I have another problem/bug :) If i restart Asterisk, the VoIP phones won't dial anything SIP related, only isdn numbers |
| 18:51.20 | Kandinsky | for a minute or so |
| 18:51.50 | [TK]D-Fender | Kandinsky: maybe becasue the phones haven't all reconnected with * because of the restart. |
| 18:51.55 | Kandinsky | I have asterisk running as user/group |
| 18:52.05 | Kandinsky | but when i had asterisk running on root account |
| 18:52.09 | Kandinsky | there was no problem |
| 18:52.14 | Kandinsky | any ideas? |
| 18:52.26 | Kandinsky | because it doesn't seem to be phone-related |
| 18:58.48 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 18:59.12 | hmmhesays | when did gmail jump to 5 gig |
| 19:00.22 | *** join/#asterisk joeballsonya (n=joeballs@adsl-69-237-115-101.dsl.scrm01.pacbell.net) |
| 19:01.15 | J4k3 | who cares |
| 19:02.57 | J4k3 | whats 5 gigs now, $1.70 worth of hard drive space? |
| 19:03.31 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 19:03.34 | J4k3 | yep, taking the price I saw for 1TB SATA HDs yesterday ($329) that works out to right at $1.70 |
| 19:04.17 | *** join/#asterisk l0verb0y (n=l0verb0y@210.1.137.41) |
| 19:04.44 | J4k3 | you let me handle your personal info, like email, and I'll be happy to give you a lot more than 5 ghz... ;) |
| 19:04.47 | J4k3 | err |
| 19:04.50 | J4k3 | 5 gb |
| 19:05.47 | l0verb0y | hey hows everyone doing |
| 19:08.49 | [TK]D-Fender | l0verb0y: jUST WORKING FOR THE WEEK-END.... |
| 19:08.56 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 19:09.11 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 19:09.29 | l0verb0y | ohhh |
| 19:09.59 | [TK]D-Fender | I R FUNNEH |
| 19:12.36 | l0verb0y | does anyone know a good forumla to figure out how many callers a machine can hold before the call quality starts going down |
| 19:13.18 | [TK]D-Fender | l0verb0y: usually more a factor of bandwidth |
| 19:13.38 | l0verb0y | assuming bandwidth was infinite |
| 19:13.39 | [TK]D-Fender | l0verb0y: Whats you server have and what about your clients? |
| 19:14.02 | [TK]D-Fender | l0verb0y: Pretty big. What kind of volume do you have in mind? And transcoding, etc? |
| 19:14.26 | l0verb0y | g729 codec, pentium 4 3.0 |
| 19:14.46 | Darthclue | hmmhesays, recently. it's been increasing steadily over time. |
| 19:15.37 | coppice | well, if you have infinite bandwidth, G.729 is a poor choice :-) |
| 19:15.44 | *** join/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 19:15.55 | *** part/#asterisk neverblue (n=neverblu@unaffiliated/neverblue) |
| 19:16.18 | l0verb0y | coppice: if you had a pentium 4 3.0 with inf bandwidth what codec would you use and how many callers could you handle |
| 19:16.21 | l0verb0y | oh also 1gb of ram |
| 19:16.24 | [TK]D-Fender | l0verb0y: indeed, and you did not actually answer my question. |
| 19:16.49 | l0verb0y | sorry |
| 19:17.14 | coppice | with infinite bandwidth, I'd use 48k samples/s PCM, 24 bits. probably stereo for good conferencing |
| 19:17.54 | [TK]D-Fender | coppice: 96k with full EAX ;) |
| 19:18.21 | [TK]D-Fender | coppice: So we can still give them that "trapped in a tin can" feeling ;) |
| 19:18.31 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 19:18.32 | coppice | yeah, surround is good for conferencing. might as well throw in HD video too |
| 19:18.48 | [TK]D-Fender | coppice: Now with extra crack! |
| 19:21.05 | l0verb0y | whats the best way to test the number of concurrent calls? |
| 19:23.46 | [TK]D-Fender | l0verb0y: "show channels concise" |
| 19:23.56 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 19:24.55 | *** join/#asterisk |omni| (n=rob@mail.argus-search.com) |
| 19:27.20 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 19:28.36 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 19:30.19 | De_Mon | I wonder if anyone has created a dialplan > flowchart script yet |
| 19:30.28 | dijungal | hello.. is there a channel variable that will show me who completed the call COMPLETEAGENT or COMPLETECALLER? |
| 19:31.01 | De_Mon | hmm |
| 19:31.13 | l0verb0y | thanks |
| 19:31.22 | De_Mon | dijungal how is the call being made? |
| 19:31.36 | dijungal | Dial |
| 19:31.59 | dijungal | Dial(SIP/${EXTEN}@provider) |
| 19:32.20 | dijungal | Dial(SIP/${EXTEN}@provider|g) |
| 19:33.32 | dijungal | i'm trying to do my own queue_log entry after the dial cmd |
| 19:33.44 | De_Mon | DIALSTATUS will be available on each side of the call leg (both channels)... maybe they will be different, worth looking into |
| 19:36.31 | dijungal | ?? |
| 19:37.24 | dijungal | how do you separate the DIALSTATUS on diff. side/leg of the call? |
| 19:37.40 | dijungal | i thought the DIALSTATUS was the status of the dial command |
| 19:38.13 | [TK]D-Fender | dijungal: it is, and De_Mon is just a little off on that. |
| 19:38.29 | [TK]D-Fender | dijungal: You'll have to account for "h" as well. |
| 19:38.54 | dijungal | k.. i was getting woried for a moment there.... i was about to say that Asterisk Bootcamp course tutor lied to me!.. lol |
| 19:39.14 | [TK]D-Fender | dijungal: "g" means COMPLETECALLER, "h" would be "COMPLETEAGENT" |
| 19:39.34 | dijungal | where do i get the g and h ? |
| 19:41.16 | [TK]D-Fender | dijungal: "g" for the dial options, "h" for the Asterisk Standard Extension. And if you have any trouble folloing this thought, then your ABC Tutor should kick you in the ass :p |
| 19:41.51 | dijungal | lol... ok i remember those now :) |
| 19:41.54 | dijungal | non need for that kick |
| 19:43.43 | [TK]D-Fender | dijungal: first one's free! |
| 19:44.02 | dijungal | [TK]D-Fender: stop the violence .... |
| 19:44.20 | [TK]D-Fender | dijungal: Ok, Homicide it is! |
| 19:44.43 | dijungal | :-/ |
| 19:44.49 | dijungal | :-! |
| 19:47.06 | *** join/#asterisk Trionnis (n=blah@209.201.67.250) |
| 19:49.39 | Trionnis | can someone point me toward some resources about the res_snmp module for 1.4.x ? I can't seem to get it to compile, even though I have all of the net-snmp packages and other related crap installed on the system. |
| 19:53.20 | *** join/#asterisk muiro (n=muiro@cpe-76-189-84-108.neo.res.rr.com) |
| 19:56.26 | *** join/#asterisk DataCompBoy (n=datacomp@213.187.250.34) |
| 19:56.57 | dijungal | u can use the 'g' parameter on the dial() command to execute more commands in the current context if the called part hangs up. what if the calling party hangsup? |
| 19:57.08 | dijungal | is there a parameter for this? |
| 19:57.46 | DataCompBoy | Hi all! Have anybody reach problem, when dialplan correctly handle DTMF, but FastAGI script using GetData don't see DTMF on SIP from SoftPhone :( |
| 19:57.57 | [TK]D-Fender | dijungal: Didn't we just go over this?! |
| 19:58.05 | dijungal | nope |
| 19:58.12 | [TK]D-Fender | dijungal: "h"!@!@ |
| 19:58.17 | dijungal | explain again |
| 19:58.25 | [TK]D-Fender | ~osmosis |
| 19:58.25 | jbot | from memory, osmosis is the act of beating yourself on the head repeatedly with THE BOOK, until some measure of absorption has occured ... or at least until your unconsciousness restores peace to the channel ... |
| 19:58.26 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 19:58.37 | Trionnis | lmao |
| 19:58.48 | Trionnis | you just got served |
| 19:58.55 | [TK]D-Fender | dijungal: When YOU hangup, the "h" ASTERISK STANDARD EXTENSIONS gets called. Use your imagination..... |
| 19:59.00 | Trionnis | don't feel bad, I get it about once a week from him |
| 19:59.03 | Trionnis | ^^ |
| 19:59.07 | Assid | anyone have ipkall forwarding the calls directly to your box and not via FWD ? |
| 19:59.08 | [TK]D-Fender | Trionnis: Hardly! |
| 19:59.17 | Assid | i dont see the call hit my box |
| 19:59.37 | Assid | i have verbose on.. and still dont see any error regarding this |
| 19:59.48 | [TK]D-Fender | Assid: [general] allowguest=yes , context=somewhere . Enable SIP debug, and test. |
| 19:59.51 | destructure | ai-a[wrk]: the timing seems pretty good actually |
| 19:59.53 | dijungal | [TK]D-Fender: when the called part or the calling party hangs up? that's what i'm trying to determine... |
| 19:59.54 | [TK]D-Fender | Assid: SIP DEBUG <------ |
| 20:00.26 | [TK]D-Fender | dijungal: "g" accounts for when the CALLEE hangs up. |
| 20:00.37 | ai-a[wrk] | destructure: does that mean your closing the Accuate Timing Testing Department ? |
| 20:00.39 | dijungal | k |
| 20:00.42 | dijungal | thnks |
| 20:01.27 | Assid | [TK]D-Fender too many sip users.. you wouldnt happen to know by chance the ip used by ipkall ? |
| 20:01.40 | [TK]D-Fender | Assid: nop. |
| 20:02.38 | destructure | ai-a[wrk]: nope, still have more to do, need to implement now. It'd be nice if I could get better than 1s resolution from strftime, but 1s will do, as long as it's spot on. I ran 1000 calls through this piii (call= get time, wait, record total to astdb). And none of them missed. the call spool was the slow part |
| 20:04.35 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 20:06.38 | DataCompBoy | AGI debug shows "AGI Rx << GET DATA ps/CN/default/appgreet 5000 1 -- <SIP/120001-08281da0> Playing 'ps/CN/default/appgreet' (language 'en') AGI Tx >> 200 result= (timeout)" while i press 1, 2, #, * etc -- no reaction. |
| 20:07.54 | Trionnis | can someone point me toward some resources about the res_snmp module for 1.4.x ? I can't seem to get it to compile, even though I have all of the net-snmp packages and other related crap installed on the system. "configure --with-netsnmp" says that it's not installed. Help please!? |
| 20:09.45 | Trionnis | the errors --> http://pastebin.ca/775948 |
| 20:10.32 | ai-a[wrk] | checking for snmp_register_callback in -lnetsnmp... no |
| 20:10.50 | GreggB | [TK]D-Fender: A few days ago you were recommending Voxee, and I just got around to checking this page which claims they're effectively out of business: http://www.voip-info.org/wiki/view/Voxee |
| 20:10.57 | Trionnis | yes, I saw that |
| 20:11.12 | [TK]D-Fender | GreggB: No I certainly did not. |
| 20:11.16 | Trionnis | I know that's why it's failing, what I can't figure out is *why* it's failing :) |
| 20:11.30 | ai-a[wrk] | triiiple: open the configure file, and see how it detects it and question why its not detecting it. |
| 20:11.31 | Trionnis | there seems to be very little documentation about res_snmp |
| 20:13.34 | GreggB | [TK]D-Fender: Hmm, alright I thought it was you, but sorry then... I could have sworn someone said that Voxee and Teliax were both working well for them. |
| 20:13.58 | *** join/#asterisk saftsack (n=saftsack@pD9E0741C.dip.t-dialin.net) |
| 20:15.08 | ai-a[wrk] | GreggB: i cant see anything in the logs. |
| 20:15.33 | *** join/#asterisk lemanal (n=lemanal@wifi-233-27.sc07.org) |
| 20:18.05 | *** join/#asterisk FlyboySR22 (n=rsears@hq.fw.americanis.net) |
| 20:19.02 | Trionnis | ok, so somewhat related question... can anyone recommend a way to monitor the current number of calls on an asterisk machine without using the manager api ? |
| 20:19.44 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:19.53 | De_Mon | Trionnis snmp |
| 20:19.56 | Trionnis | ... |
| 20:20.04 | Trionnis | ...... |
| 20:20.10 | ai-a[wrk] | Heh. |
| 20:20.29 | De_Mon | I suggest you get it working |
| 20:21.14 | Trionnis | yes, I'll recode the configure script... I'll get right on that |
| 20:21.32 | Trionnis | oh wait, I know jack shit about those... maybe that's not such a good idea :) |
| 20:21.35 | De_Mon | you assume its the configure script at fault |
| 20:21.58 | ai-a[wrk] | the configure script is telling you something is wrong on your box. |
| 20:22.27 | De_Mon | failing that you could always hire something that does know what they are doing ^_^ (not me) |
| 20:22.36 | ai-a[wrk] | "The Net-SNMP installation on this system appears to be broken." i would be worried about why your snmp isnt working. |
| 20:24.31 | asdx | yay, i've got the teliax account now... should i talk with my customer about the price of work, etc, before i start working? |
| 20:25.32 | De_Mon | asdx ... you should always have a contract before starting work stating exactly what your doing and how much he is going to play for it |
| 20:25.50 | De_Mon | any changes to the contract means ammending the contract including how much he is going to pay... |
| 20:26.00 | GreggB | [TK]D-Fender: Alright, sorry for the confusion. It was teliax and VoicePulse that you stated were mostly decent back on the 6th. |
| 20:26.21 | [TK]D-Fender | GreggB: better. |
| 20:26.21 | GreggB | I take it Voxee is either failing or out of business, despite the fact they'll still take your money on the website, and that they still "rank" in the top 10% on www.myvoipprovider.com Sigh... Is anyone using them right now? |
| 20:27.06 | asdx | De_Mon: i see, but... the problem is that i am a noob, and i still have to learn some things and my customer ask me "how much time will it take" and i don't know how much time will take for me to learn/implement that, but i really want to do it |
| 20:27.38 | asdx | De_Mon: i guess is not much work and i'll learn fast... i usually learn fast. |
| 20:28.12 | De_Mon | its better to under promise and over deliver than the other way around |
| 20:28.15 | [TK]D-Fender | asdx: You are clearly in way over your head and should not even be offering * services. |
| 20:28.49 | asdx | yeah, i should learn first |
| 20:28.50 | De_Mon | are you even old enough to work at mcdonalds? |
| 20:29.37 | J4k3 | asdx: you need to be trixbox certified! its only $5k USD! |
| 20:29.40 | Assid | [TK]D-Fender : i got ipkall to connect to me.. now im getting Looking for external-shahbc in inbound (domain domain.name.here) |
| 20:29.41 | J4k3 | (NOT) |
| 20:29.43 | Trionnis | lmao |
| 20:29.58 | asdx | J4k3: trixbox? why that? i prefer cli |
| 20:30.09 | Trionnis | Trixbox certified.... the MCSE of the Asterisk world |
| 20:30.12 | asdx | De_Mon: i'm 24 |
| 20:30.23 | asdx | M$ sucks |
| 20:30.32 | rob0 | Two dozen. |
| 20:30.35 | De_Mon | just wow |
| 20:30.41 | rob0 | 42 inverted |
| 20:31.02 | J4k3 | asdx: once you get past 'what sucks' you might learn something. |
| 20:31.10 | asdx | well, i'll do this for free... and i'll use this experience to learn. |
| 20:31.10 | J4k3 | but, when I was 24 my anger was a lot less focused, I understand. |
| 20:32.02 | asdx | J4k3: anger? |
| 20:32.09 | J4k3 | today I'm quite a bit more upset about, say, scorching my lunch noodles, than I am pointing at some 'successful' company and saying they suck. |
| 20:32.16 | [TK]D-Fender | J4k3: Time to let go of anger and move on to full blown psychosis! ;) |
| 20:32.26 | asdx | J4k3: they suck for many reasons. |
| 20:33.08 | J4k3 | asdx: don't knock the dopeman's hustle. |
| 20:33.16 | J4k3 | and thats pretty much my opinion of Microsoft... ;) |
| 20:33.19 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 20:33.37 | De_Mon | my ghetto lingo isn't up to par with that statement |
| 20:33.46 | J4k3 | if you're angry at microsoft, you need to re-evaluate and figure out a way to profit off the situation. |
| 20:34.02 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 20:34.08 | De_Mon | oh, okay |
| 20:34.15 | J4k3 | Microsoft sells crappy products that screw people up to always being Microsoft-users. No different than dope. |
| 20:34.25 | Trionnis | or Cisco |
| 20:34.27 | Assid | anyone know where "425" is used ? |
| 20:34.31 | Assid | as in the area code |
| 20:34.35 | rob0 | WA |
| 20:34.37 | Trionnis | um.. after 424? |
| 20:34.41 | J4k3 | but, when the local dope dealer drives through in a $100k car, its hard to deny the business plan's workability. |
| 20:34.42 | De_Mon | you should try google |
| 20:35.29 | Assid | yeah.. was referring to the ipkall numbers :P |
| 20:35.51 | rob0 | http://www.nanpa.com/area_codes/ |
| 20:36.10 | De_Mon | J4k3 bad business practices does not a good business make. Evil always has its hour, day, year... but will eventually reap what they sow |
| 20:36.23 | Trionnis | whoa... that's deep |
| 20:37.06 | asdx | [TK]D-Fender: this guy (customer) told me that he will give me some time to learn |
| 20:37.39 | [TK]D-Fender | asdx: desperate chump. Sounds like a dangerous cutsomer. |
| 20:37.50 | asdx | [TK]D-Fender: i have my computer here and i compiled/installed asterisk, but i have no hardware and no voip provider... |
| 20:38.03 | *** join/#asterisk bantu (n=Miranda@p54A316EC.dip0.t-ipconnect.de) |
| 20:38.06 | asdx | [TK]D-Fender: they are giving me a voip provider now for playing... |
| 20:38.11 | [TK]D-Fender | asdx: 1 monkey short of a full set ;) |
| 20:38.26 | J4k3 | so they're investing $1.50/month and 1.1 cents per minute into your success ;) |
| 20:38.30 | J4k3 | cheap investment, I must say. |
| 20:38.33 | De_Mon | rob0 where is the actual list of areacodes there? |
| 20:38.39 | Assid | whats wrong with ipkall with free incoming? |
| 20:38.53 | Assid | J4k3 ^ |
| 20:39.01 | asdx | hmm |
| 20:39.13 | De_Mon | omg an access database? kill me now |
| 20:39.22 | Assid | where where? |
| 20:39.34 | J4k3 | ax-sex. |
| 20:40.25 | Assid | J4k3 thats what it feels like once it crosses 28MB |
| 20:41.07 | J4k3 | haha |
| 20:41.28 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 20:41.31 | J4k3 | its acceptably bad on ethernet. |
| 20:41.43 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 20:43.14 | De_Mon | Aww they have a city/county lookup tool, I want that |
| 20:47.03 | *** join/#asterisk Steven_elvisda_ (n=Steven_E@202.47.107.60) |
| 20:48.51 | hmmhesays | anyone using broadvoice with asterisk 1.4? |
| 20:50.19 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 21:06.04 | Trionnis | not if they're smart |
| 21:06.16 | Trionnis | broadvoice is an evil, evil company |
| 21:07.31 | J4k3 | I've found anything with 'broad' in it should be avoided. |
| 21:07.41 | J4k3 | broadcom, broadwing, broadvoice, etc. |
| 21:07.56 | J4k3 | I prefer chicks, avoid the broads. |
| 21:09.15 | Trionnis | broadwing..... *growl* |
| 21:09.33 | Trionnis | "oh, here's your NI2 PRI" "uh, you're passing 'plan: private' on blocked CID calls... that's not following the standard" "oh, well whatever, deal with it" |
| 21:09.54 | Trionnis | yeah, fuck Broadwing... fuck Broadwing sideways with a broomhandle... |
| 21:10.33 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 21:11.58 | *** join/#asterisk BBHoss (n=hoss@146.229.191.76) |
| 21:12.13 | [TK]D-Fender | J4k3: Broadway has made me a fortune on Monopoly ;) |
| 21:13.14 | Qwell | [TK]D-Fender: it's really Park Place |
| 21:13.21 | Qwell | Broadway is just a freeloader |
| 21:13.27 | *** join/#asterisk kingsob (n=derek@TOROON01-1177844188.sdsl.bell.ca) |
| 21:13.29 | BBHoss | heh |
| 21:13.43 | kingsob | does anyone know anyhting about unlimitel? |
| 21:14.19 | kingsob | im wondering if its possible to get a number with the ability to have like 30 imbound calls as the same time |
| 21:15.00 | [TK]D-Fender | kingsob: their base is 5 calls / DID, but you can call them up and arrange something bigger. |
| 21:15.06 | [TK]D-Fender | kingsob: like VPRI, etc. |
| 21:15.46 | [TK]D-Fender | kingsob: then again at that many channels, and for the hassle & risks you might be better of getting your own PRI... |
| 21:16.17 | kingsob | I kinda want to offer a service to my friends where they can call my number |
| 21:16.22 | kingsob | and make free outgoing calls |
| 21:16.26 | kingsob | to longdistance numbers |
| 21:16.28 | *** join/#asterisk galeras (n=Martin@201.244.247.149) |
| 21:16.36 | kingsob | by dialing a local number |
| 21:17.38 | kingsob | how much does a PRI cost? |
| 21:18.44 | outtolunc | more than the 'free' you'll be charging your friends <G> |
| 21:19.41 | outtolunc | want fries with that? <G> |
| 21:19.43 | [TK]D-Fender | kingsob: Depends where you are and what provider you're looking at |
| 21:20.02 | kingsob | well maybe not free, just a small charge to cover my costs |
| 21:20.14 | BBHoss | are you in a metro area? |
| 21:20.20 | kingsob | yeah, toronto |
| 21:20.28 | [TK]D-Fender | kingsob: at that rate they may as well just set themselves up with SkypeOut and be done with it. |
| 21:20.43 | [TK]D-Fender | kingsob: And save yourself a boatload of $ |
| 21:20.49 | [TK]D-Fender | kingsob: And troubble |
| 21:20.56 | BBHoss | im not sure about canada, probably anywhere from 300-600USD a month |
| 21:20.58 | kingsob | yeah, but u cant use skype to call a local number on ur cellphone |
| 21:21.12 | lirakis | how do i determine the "return code" from a carrier? .. so i can play a different sound if it is busy 17, or some thing else for another code? is it stored in a chan var? |
| 21:21.33 | kingsob | thats cool, even at like 2000/mo in costs, i would only have to sell it to like 50 people @ 20/mo to cover costs |
| 21:21.51 | J4k3 | 100 people at 20/month |
| 21:21.55 | J4k3 | makes $2000. |
| 21:22.01 | kingsob | lol yes, yes it does |
| 21:22.20 | J4k3 | but |
| 21:22.21 | kingsob | but still, that should be relatively easy |
| 21:22.31 | J4k3 | a PRI in the GTA will run you max $250/month |
| 21:22.38 | kingsob | the likely hood of them all calling in at exact same time is quite small |
| 21:22.39 | J4k3 | but that'll require a colo to get a price like that |
| 21:22.52 | kingsob | colo? |
| 21:22.53 | J4k3 | and last time I paid sub-$250 for PRIs, a rack was $750/month |
| 21:22.55 | J4k3 | colocation |
| 21:23.07 | J4k3 | basically putting your gear at the switch, rather than them bringing a circuit to you. |
| 21:23.10 | [TK]D-Fender | J4k3: 250$ for GTA? Thats crazy low.... |
| 21:23.11 | Assid | damn. i wish freecall/voipstunt etc. allowed callerid |
| 21:23.29 | Assid | atleast they have free sms for a bit |
| 21:23.32 | [TK]D-Fender | J4k3: Best I get here in Mtl its $650 |
| 21:23.54 | kingsob | im just trying to come up with a buisness idea using asterisk |
| 21:24.00 | J4k3 | [TK]D-Fender: $250 is crazy low anywhere, its just a matter of finding it. hell, I was paying $210/month per PRI and like $0.10/DID/month to Focal |
| 21:24.09 | kingsob | ever since i started playing around with it about a year ago, i;ve fell in love |
| 21:24.22 | kingsob | amazing what u can do with it |
| 21:24.23 | J4k3 | of course the service went to shit, and they attempted billing me double, when they were bought out by broadwing (hence 'avoid anything with broad in its name') |
| 21:24.45 | galeras | Hi Sirs, my client is reporting this weird behavior: Occasionally, same call is incoming for 2 diferent ZAP channels (i.e. 27 and 84). Any idea what can be happening? |
| 21:24.46 | BBHoss | yeah pris are still damn high around here |
| 21:25.05 | [TK]D-Fender | kingsob: Then open a phone-sex line like the rest of us..... |
| 21:25.21 | [TK]D-Fender | kingsob: Leave termination out of it ;) |
| 21:25.27 | kingsob | lol |
| 21:26.11 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 21:26.34 | kingsob | maybe ill write an ai, and allow people to call in for phone sex with my computer voice, thatd be sexy |
| 21:28.10 | J4k3 | get a PRI |
| 21:28.16 | J4k3 | and some old Apple IIs |
| 21:28.25 | J4k3 | and find a copy of diversi-dial |
| 21:28.28 | BBHoss | haha |
| 21:28.43 | [TK]D-Fender | ok, time to head home. BBIAB |
| 21:29.18 | creativx | you could probably earn millions on it |
| 21:30.15 | BBHoss | yeah me and a friend actually thought of that, but we decided the computers still aren't powerful enough to UNDERSTAND voice 100%, much less speak fluently |
| 21:31.37 | J4k3 | shit, I just remembered I can dial anywhere in the USA same as local on my cellphone, up to 14.4k |
| 21:32.13 | BBHoss | heh |
| 21:32.20 | BBHoss | use the 10-228 trick to spoof your ANI :) |
| 21:32.32 | BBHoss | or 10-10-228 |
| 21:33.15 | J4k3 | I did that, and ended up getting a $1k+ phone bill from metromedia a few months later :E |
| 21:33.28 | Qwell | J4k3: that's the whole use of that number |
| 21:33.28 | BBHoss | fun |
| 21:33.29 | Qwell | :p |
| 21:33.46 | J4k3 | yeah, I mowed a lot of yards... :/ |
| 21:39.51 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
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| 21:42.18 | fbnts | Hi, is there a way to log on an agent from a script? I have looked at the CLI but it only allows you to logoff or show agents |
| 21:44.23 | asdx | damn this customer is desperate for his pbx, but he doesn't want to pay me... |
| 21:44.35 | asdx | and he is pressuring me, i don't like this |
| 21:45.18 | Maliuta | I thought it was simple: no payee, no workee |
| 21:45.55 | asdx | Maliuta: i want to learn how to connect asterisk with a voip provider :-) |
| 21:46.31 | asdx | but maybe i should get my own account for that... |
| 21:46.43 | Maliuta | asdx: you don't have to work for someone to do that |
| 21:47.06 | asdx | Maliuta: yeah |
| 21:47.21 | Maliuta | asdx: I am going to get to do some asterisk stuff in my new job, but all the stuff I have learnt has been on my own kit |
| 21:48.06 | Maliuta | beneficial to me too, given that I have been living 2000kms from most of my friends and my parents live in canada |
| 21:48.35 | Maliuta | $0.08 calls to any landline in .au, .uk, .ca ..... |
| 21:49.40 | asdx | Maliuta: is there free voip providers that i can use? |
| 21:49.50 | asdx | Maliuta: for learning purposes |
| 21:50.01 | Maliuta | there fwd |
| 21:50.17 | Maliuta | I pay for my DID and any calls I make |
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| 21:57.18 | Trionnis | hooray for netsplits |
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| 22:01.43 | jameswf | so neat whena a node poops |
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| 22:01.44 | Qwell | that had to have been more than a single node |
| 22:01.45 | muiro | that was 4 nodes by my count |
| 22:01.45 | muiro | right, I'll repeat my question now: |
| 22:01.45 | muiro | question: what do I need to modify to make saynumber, saydigits, etc. use background() instead of playback()? |
| 22:02.08 | muiro | if it's possible |
| 22:02.43 | Qwell | it's not |
| 22:02.43 | Qwell | or, rather, not currently |
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| 22:02.51 | muiro | hrat |
| 22:02.55 | muiro | drat, even |
| 22:03.00 | Maliuta | muiro: just write your own macro to use background to play the files |
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| 22:07.39 | muiro | yeah, that's whait I'm going to do Maliuta, easy enough |
| 22:09.19 | rob0 | This channel is registered users only, right? |
| 22:09.27 | Qwell | yes |
| 22:09.46 | rob0 | So when I get dumped by a netsplit, my window is lost and I have to manually rejoin. |
| 22:10.20 | rob0 | Have you guys considered just taking away voice from unregistered users? |
| 22:11.06 | Qwell | I don't think there's a flag for that |
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| 22:17.41 | [TK]D-Fender | (S)Qwell(tched)! |
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| 22:21.51 | Dovid | is switchvox based on asterisk ? |
| 22:24.26 | hmmhesays | the mind is a terrible thing to waste |
| 22:26.27 | hmmhesays | broadvoice is a bunch of b1tches |
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| 22:29.37 | Corydon76-vcch | Dovid: yes |
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| 22:33.36 | teknoprep | where is the asterfax config file? |
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| 22:57.02 | killfill | hi.. |
| 22:57.18 | killfill | i cannot get something from my dialplan |
| 22:57.22 | killfill | http://pastebin.ca/776208 |
| 22:57.42 | killfill | when i call from extension 45 to 31, the macro "macro-stdexten" gets executed |
| 22:58.00 | killfill | but from 45 to 23, i get a plain Dial("IAX2/45-1", "SIP/23&IAX2/23") |
| 22:58.05 | *** join/#asterisk johndbritton (n=john@cpe-74-70-255-158.nycap.res.rr.com) |
| 22:58.07 | killfill | why is it different? |
| 22:58.44 | johndbritton | im getting a registration failed for xxx.xxx.xxx.xxx no matching peer found how can i make the peer match? |
| 22:58.55 | johndbritton | im trying to connect from a hardphone |
| 23:00.27 | killfill | i have nothing specific to 23 or 31 in the whole dialplan. |
| 23:00.40 | killfill | why do i get different behaviour?.. |
| 23:07.08 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 23:12.34 | killfill | damn.. :S |
| 23:18.13 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 23:19.28 | killfill | oh.. |
| 23:19.29 | killfill | wired.. |
| 23:19.37 | killfill | its becouse one user has voicemail and the other not. |
| 23:19.39 | killfill | hm.... |
| 23:20.55 | killfill | when the user has voicmail, the std-exten get called.. |
| 23:21.05 | killfill | where is this controlled?.. how do i change this behaviour? |
| 23:25.23 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:25.59 | *** join/#asterisk slowshutt (n=d@196.211.34.2) |
| 23:27.30 | *** join/#asterisk ZenBSDi (n=zenbsdi@unaffiliated/ZenBSDi) |
| 23:34.27 | fujin | killfill: a local channel will run through the dialplans and guess how to best contact the dialee |
| 23:35.02 | *** join/#asterisk Zuchmir (n=dddddd@123-2-65-142.static.dsl.dodo.com.au) |
| 23:36.29 | Zuchmir | how can i tell if my digium card is working |
| 23:36.30 | slowshutt | if one has a asterisk box on a static ip linking it to the internet where can one find info on possible security risks? |
| 23:38.38 | killfill | fujin: ok thats fine. my problem is, that i need to execute something before a phone gets ring. so i was putting things in stdext macro. This works. but for users that has voicmails... |
| 23:39.04 | killfill | for users that doesnt (i.e. users that are part of a Queue), how would i catch them? |
| 23:39.13 | fujin | queue is entirely different |
| 23:39.30 | fujin | the only way I've been able to specifically do things for queue members, is to write a dynamic queue member system |
| 23:39.32 | fujin | with a Local channel |
| 23:39.44 | fujin | (I did it in AEL, was probably 2-3 hours work to get it perfect) |
| 23:39.47 | killfill | hm... |
| 23:41.10 | killfill | fujin: is it possible that a user has voicmail, and he can also exist in a queue as agent?. currently, im taking hes voicmail out, becouse when the queue calls him, and is busy, voicmail apears, and not the next agent.. |
| 23:43.49 | *** join/#asterisk khronos (n=khronos@c-66-229-159-201.hsd1.fl.comcast.net) |
| 23:44.03 | asdx | i already got a ITSP (teliax) and i have a server with asterisk already... i want to connect to my asterisk and then make outgoing calls |
| 23:44.15 | *** join/#asterisk angom (n=angom@200.38.31.239.dsl.dyn.telnor.net) |
| 23:44.30 | asdx | teliax sent to me a user configuration with: |
| 23:44.30 | slowshutt | yes asdx are you using sip or iax2? |
| 23:44.32 | asdx | host=voip-co4.teliax.com |
| 23:44.36 | asdx | slowshutt: yes, iax2 |
| 23:44.53 | slowshutt | can you not connect? |
| 23:45.03 | asdx | slowshutt: i can connect if i set host=dynamic |
| 23:45.19 | asdx | slowshutt: but that host=voip-co4.teliax.com confuses me |
| 23:45.20 | Zuchmir | i have 1 FXO, i plugged it in, followed the books "minimal script" (changing 2 to 1) and when i call in, it does not answer |
| 23:46.26 | slowshutt | just ask them to add your ip address to their iax.conf then ip will work they keep theirs dynamic because they don't lnow where you connecting from |
| 23:47.08 | johndbritton | ive got a nat problem i think... ive got a hardphone on my internal (home) network conncected via internet to my asterisk box (on the net with public ip) I am able to place a call from my hardphone, and i can hear the person im calling but they cannot hear me... any suggestions |
| 23:47.23 | slowshutt | thant means you are connectin to there ip hostname =voip-co4.teliax.com |
| 23:48.01 | asdx | slowshutt: so i need two [users] entries? |
| 23:48.29 | slowshutt | paste your iax.conf and i will have a look remember to xxxxx out your password |
| 23:48.29 | slowshutt | no |
| 23:48.32 | slowshutt | just one |
| 23:48.58 | *** join/#asterisk stubert (n=stu@techtools.actusa.net) |
| 23:49.25 | slowshutt | iax is a bothway connection you set up your iax to connect to them and visa versa |
| 23:51.09 | asdx | slowshutt: http://pastebin.ca/776277 |
| 23:51.54 | slowshutt | asdx link not active |
| 23:51.56 | asdx | slowshutt: do i need another host so my client here in my computer will be able to connect? |
| 23:52.37 | slowshutt | pastebin link dead |
| 23:52.42 | asdx | wait |
| 23:53.11 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 23:53.53 | slowshutt | use pastebin.com |
| 23:54.20 | asdx | http://pastebin.com/m785a5959 |
| 23:55.46 | slowshutt | you dont need the register string |
| 23:56.19 | asdx | ok |
| 23:56.38 | slowshutt | what do you get if you type iax2 show peers from asterisk console? |
| 23:57.21 | johndbritton | the person im calling from my sip hardphone cant hear me, but i can hear them |
| 23:57.56 | asdx | slowshutt: diego 63.211.239.2 (S) 255.255.255.255 4569 Unmonitored |
| 23:58.05 | slowshutt | add username=diego and secret=foobar i your diego context in iax.conf |
| 23:58.23 | *** join/#asterisk r0d3nt (n=astrutt@foster.stonedcoder.org) |
| 23:58.45 | asdx | ok |
| 23:59.23 | asdx | done |
| 23:59.50 | slowshutt | what do use to dial use the line for outgoing |
| 23:59.58 | slowshutt | i presume you have a sip phone |