| 00:00.08 | mackes | Well, my corp uses them- All over |
| 00:00.13 | drmessano | So? |
| 00:00.16 | mackes | So does many supermarkets |
| 00:00.24 | drmessano | Wifi handhelds suck.. period |
| 00:00.25 | mackes | And Hospitals |
| 00:00.33 | *** join/#asterisk CpuID (i=zkodeh@gentoo/contributor/cpuid) |
| 00:00.36 | mackes | THAT IS NOT TRUE!!! It just isnt |
| 00:00.49 | mackes | It all depends on your WiFi Network |
| 00:00.51 | JT | OMG LETS SHOUT LIKE AN IDIOT |
| 00:01.00 | JT | yes if you spend a lot on your wifi network, it sucks a lot less |
| 00:01.02 | drmessano | It has nothing to do with the NETWORK |
| 00:01.04 | CpuID | hey ppls, anyone here pretty familiar with PRI? im wanting to try set something up as a form of PRI emulator to test E1 connectivity between 2 devices |
| 00:01.08 | mackes | http://www.voipsupply.com/product_info.php?products_id=354 |
| 00:01.16 | CpuID | i gather its not the same as just FXOs talk to FXSs like with analog :P |
| 00:01.17 | drmessano | Sure, network can DEGRADE IT |
| 00:01.19 | mackes | This phone works |
| 00:01.26 | drmessano | But wifi sip phones suck |
| 00:01.48 | mackes | And I am beta testing a Polycom Spectralink right now that is very good as well |
| 00:01.54 | JT | CpuID: 2 ports on an e1 card? |
| 00:02.38 | CpuID | 2 ports? |
| 00:02.50 | CpuID | can e1 ports just talk to eachother? or is there like a headend/CPE type scenario? |
| 00:02.56 | JT | sure |
| 00:02.58 | CpuID | oh... |
| 00:03.07 | JT | set one to pri_net one to pri_cpe, use a crossover cable |
| 00:03.11 | CpuID | so its not like 1 provides voltage and the other just listens... |
| 00:03.12 | CpuID | o0 |
| 00:03.13 | JT | set correct timing in zaptel.conf |
| 00:03.14 | CpuID | interesting :) |
| 00:03.27 | CpuID | this i didnt know :) |
| 00:03.27 | mackes | How can you make a declarative statement like "wifi sip phones suck" .. How could you possibly know that for sure? Have you had so much experience with SIP Wifi roll out that you are judge of a technology |
| 00:03.43 | CpuID | any particular pci cards required to support pri_net? |
| 00:03.48 | JT | no |
| 00:03.52 | CpuID | i gather pri_cpe would be supported by practically anything |
| 00:03.53 | CpuID | nice one :) |
| 00:04.08 | CpuID | woo this just saved me some hassles lol |
| 00:04.11 | drmessano | I have had experience with some of them, and you are getting way too emo over this |
| 00:04.48 | CpuID | now all i need to do is find the cheapest E1 pci option i can :) |
| 00:05.07 | CpuID | wants to try get a cisco AS5300 access gateway to talk to an asterisk box as a prototype |
| 00:05.25 | CpuID | then once all is good, replace the asterisk box with some telco E1s |
| 00:06.22 | JT | heh |
| 00:06.53 | JT | make sure you use an e1 crossover cable |
| 00:07.01 | Zorix | is there any reason why aserisk system voices such as dialing into conference number are really choppy and lagged, while the music seems to work? |
| 00:07.19 | drmessano | bandwidth |
| 00:07.21 | mackes | All I am saying is that it doesnt help Open Source VoIP to shoot down WiFi SIP on the most important IRC channel for Asterisk discussion. Some of the manufactures who develop these products visit this room. I would hate to think that we are scaring away customers from their products towards Cisco, Nortel and Alcatel WiFi, non SIP products |
| 00:07.30 | JT | could be zap timing, Zorix |
| 00:07.31 | CpuID | thanks for the advice there JT :) |
| 00:07.31 | jbot | CpuID: my pleasure |
| 00:07.33 | CpuID | ill make sure of that |
| 00:07.37 | CpuID | hahahaha |
| 00:07.40 | CpuID | damn bawt |
| 00:07.45 | Zorix | JT, how do i adjust that |
| 00:07.57 | JT | get a zaptel card |
| 00:08.24 | Zorix | zttest showes like -200% |
| 00:08.31 | Zorix | why, i am only using sip |
| 00:08.32 | JT | mackes: it would be great if we scared them towards dect sip solutions |
| 00:08.36 | dachary | mackes: http://www.voipsupply.com/product_info.php?products_id=2996 is not available. Would you recommend another hitachi product ? |
| 00:08.42 | JT | Zorix: meetme is zaptel conferencing |
| 00:08.46 | drmessano | As soon as someone makes a decent wifi chipset that doesn |
| 00:08.53 | Zorix | is there a software solution? |
| 00:09.06 | mackes | Sorry- it was upgraded |
| 00:09.09 | drmessano | As soon as someone makes a decent wifi chipset that doesn't drain batteries, and then puts it in a phone that doesn't sound like crap, then you'll have something |
| 00:09.09 | mackes | http://www.voipsupply.com/product_info.php?products_id=2996 |
| 00:09.13 | JT | Zorix: you must already have ztdummy otherwise it wouldn't already work |
| 00:09.16 | JT | Zorix: but no |
| 00:09.24 | JT | Zorix: otherwise there's app_conference |
| 00:09.25 | drmessano | Until then, as JT said, DECT or a standard cordless with an ATA is far better |
| 00:09.26 | Zorix | yes ztdummy |
| 00:09.31 | *** join/#asterisk thepacmanfan (n=thepacma@12-218-140-147.client.mchsi.com) |
| 00:09.41 | jameswf-home | yes if an oem comes here and sees that everyone thinks they suck maybe they will improve |
| 00:09.49 | JT | wifi will always drain batteries |
| 00:09.55 | dachary | mackes: upgraded, yes. But the upgrade is no available at the moment. |
| 00:09.57 | JT | and there's still fundamental issues |
| 00:10.03 | mackes | But those solutions do not cover Campus wide networks- or Corp networks that cover multiple locations |
| 00:10.20 | jameswf-home | pandering for broken goods wtf |
| 00:10.22 | JT | unless your wifi infrastructure is top notch, you will have issues with mobile wifi voip |
| 00:10.44 | thepacmanfan | ok, new asterisk user here... i have a phone set up and seemingly communicating with my Asterisk server, but whenever i try and place a call i get a fast busy signal. |
| 00:11.08 | drmessano | If someone comes in here and hears that WIFI SIP phones suck, so they go out and spend 10x as much on a Cisco, then they have issues |
| 00:11.08 | Zorix | JT, would this option help: internal_timing=yes ? |
| 00:11.15 | drmessano | "ZOMG, they scared me to Lucent" |
| 00:11.16 | jameswf-home | Its like politicians in AZ pandering to a group because of their numbers even though they cant vote... the math isnt there |
| 00:11.18 | drmessano | Uh, no |
| 00:11.18 | thepacmanfan | i'm using an FXO card for incoming/outgoing analog lines |
| 00:11.28 | JT | Zorix: where is that option? |
| 00:11.35 | Zorix | i found it in a search |
| 00:11.42 | Zorix | web search |
| 00:11.44 | mackes | Here is the more consumer version of the Business Hitachi |
| 00:11.47 | mackes | http://www.ipphone-warehouse.com/Hitachi-Cable-WirelessIP3000-Hitachi-IP-3000-WiFi-p/hitachi-wireless%20ip3000.htm |
| 00:11.48 | thepacmanfan | it seems like Asterisk is giving the busy signal, not the analog line. |
| 00:11.59 | JT | Zorix: err great. what config file? |
| 00:12.11 | Zorix | says asterisk.conf |
| 00:12.17 | thepacmanfan | maybe my dial plan isn't properly set up? |
| 00:12.28 | drmessano | Fact is, WIFI SIP phones are a solution for a problem is based on inherently flawed tech anyway |
| 00:12.40 | mackes | Well, here is my take. We have about 200 SIP WiFi phones deployed Zultys WIP2- |
| 00:12.46 | mackes | They all work very well |
| 00:12.54 | mackes | As well as a cell phone |
| 00:12.56 | Zorix | JT, will let you know if it works |
| 00:12.57 | drmessano | You use all 200 of them? |
| 00:12.59 | drmessano | Every day? |
| 00:13.01 | mackes | yes |
| 00:13.05 | drmessano | YOU DO? |
| 00:13.06 | JT | how many thousands were spent on the APs? |
| 00:13.07 | drmessano | PERSONALLY? |
| 00:13.12 | drmessano | YOU use all 200 phones? |
| 00:13.13 | drmessano | WOW |
| 00:13.25 | jameswf-home | cough DRUGDEALED cough |
| 00:13.27 | mackes | we have as many as 10 AP's per location |
| 00:13.32 | jameswf-home | *dealer |
| 00:13.36 | mackes | Alcatel WiFi System |
| 00:13.49 | drmessano | I find it hard to believe you have used all 200 of those on a daily bases |
| 00:13.51 | drmessano | basis |
| 00:13.52 | mackes | 350 AP's in total (give or take) |
| 00:14.02 | JT | that's my point, you need to spend heaps on the infrastructure before the phones have a hope of working ok |
| 00:14.05 | mackes | We have 28 Retail locations |
| 00:14.06 | JT | lol 350 |
| 00:14.24 | mackes | Large Campus locations |
| 00:14.25 | thepacmanfan | they work as well as a cell phone? that's not very encouraging! |
| 00:14.30 | drmessano | Fact is, users will use the crap out of shit that sounds "OK" and you'll never hear from them unless its broken |
| 00:14.40 | drmessano | So that tells me nothing |
| 00:14.42 | jameswf-home | well a kid doesnt know a stale cookie sucks until they have one fresh baked so a wifi sip user doesnt realize how bad it sucks cuz they havent seen anything better |
| 00:14.47 | mackes | ok. you win. |
| 00:15.03 | drmessano | People use Spyware infected PCs all day and only stop clicking popups when the machine wont boot |
| 00:15.03 | thepacmanfan | i'm used to physically-linked PBX systems being orders of magnitudes more reliable than cell phones. |
| 00:15.11 | JT | wifi is not well suited to mobile voip, but you can make it work OK |
| 00:15.25 | file | hugs DECT |
| 00:15.41 | JT | wifi is lossy, jittery, and half duplex |
| 00:15.53 | JT | and subject to interference and environmental attenuation |
| 00:16.05 | mackes | So is all wifi- |
| 00:16.10 | JT | correct |
| 00:16.18 | mackes | including 9ooMhz phones |
| 00:16.19 | drmessano | thinks JT already said that |
| 00:16.26 | drmessano | 900MHZ phones are not "wifi" |
| 00:16.26 | thepacmanfan | 900mhz phones != wifi |
| 00:16.31 | drmessano | WTF |
| 00:16.45 | JT | but it's a specially big problem when doing streaming full duplex comms |
| 00:17.02 | thepacmanfan | wifi is not designed to gracefully handle momentary signal breakups. |
| 00:17.14 | JT | i'd love to see all 200 wifi phones at one location strike up a conversation at once |
| 00:17.17 | JT | *boom* |
| 00:17.24 | drmessano | or 25 of them |
| 00:17.40 | mackes | Hmmmmm... We can |
| 00:17.45 | drmessano | lol |
| 00:18.09 | drmessano | You've done it? |
| 00:18.13 | drmessano | All 200 at once? |
| 00:18.16 | drmessano | One site |
| 00:18.41 | mackes | No- 6 phones + per site, 28 retail sites |
| 00:18.49 | mackes | each site has a T1 |
| 00:18.59 | JT | we're talking 15000 pps+ switching between tx and rx and different stations |
| 00:18.59 | mackes | Each site has around 10 AP's |
| 00:19.00 | drmessano | JT said "one location" |
| 00:19.11 | thepacmanfan | so 12-13 APs per site on average, and only 6+ users? meh. |
| 00:19.13 | drmessano | and you answered "We can" |
| 00:19.50 | mackes | For coverage... I get the impression that you dont do large scale rollouts in WiFi? |
| 00:19.51 | JT | 6 phones per 10 aps |
| 00:19.54 | JT | childs play. |
| 00:20.02 | thepacmanfan | if it works for you, more power to you... but it seems vastly impractical for 99% of situations. |
| 00:20.05 | mackes | Picture a large retail complex |
| 00:20.08 | drmessano | Actually, I do |
| 00:20.29 | JT | so what is that, $10k of APs per location? |
| 00:20.31 | dFence | strrrrange: with nat=no can canreinvite=no for SIP1 and an offline SIP2 the Dial(SIP/SIP1&SIP/SIP2) fails immediately |
| 00:20.34 | mackes | 1 AP would handle an area of the building without an issue |
| 00:20.41 | mackes | ok, never mind. |
| 00:21.02 | JT | mackes: how much is a typical 10 AP delpoyment? |
| 00:21.07 | mackes | Its about $125 per AP |
| 00:21.14 | drmessano | $125? |
| 00:21.17 | drmessano | What APs? |
| 00:21.20 | JT | not installed |
| 00:21.25 | mackes | We install them |
| 00:21.32 | JT | you install them for free? |
| 00:21.48 | drmessano | What AP's are you talking about for $125? |
| 00:22.03 | JT | i'm sure APs sell for $125 or less |
| 00:22.04 | mackes | http://www1.alcatel-lucent.com/products/productsummary.jsp?relativePath=/com/en/appxml/opgproduct/omniaccesswlanfamilytcm228121901635.jhtml |
| 00:22.06 | thepacmanfan | $125 will not buy an AP i would consider using for phone service. |
| 00:22.15 | JT | i'm also sure that's not fully installed |
| 00:22.31 | thepacmanfan | Lucent sells an AP for $125? holy cow. |
| 00:23.00 | mackes | The AP's are just dumb clients to two central controllers. They can find a client down to a meter |
| 00:23.23 | mackes | Enterprise Grade folks |
| 00:23.28 | JT | so how much is the central controllers costing? and how much to install the whole shebang? |
| 00:23.31 | drmessano | We use Cisco 1131AGs |
| 00:23.44 | drmessano | I wouldnt touch a $125 AP |
| 00:23.44 | mackes | However, I use one Cisco 1200, and two SIP wifi phones in my home- no issues |
| 00:23.57 | mackes | Go to the link |
| 00:24.00 | drmessano | Why not just use a Linksys WRT54G if you're gonna spend that little |
| 00:24.09 | JT | mackes: why do you always avoid every second question? |
| 00:24.25 | JT | mackes: how much to instll everything? |
| 00:24.34 | JT | parts is only half the equation for a business |
| 00:24.39 | JT | labour is the other half |
| 00:24.57 | thepacmanfan | drmessano: only if you want to go around rebooting 350 APs every few months |
| 00:24.59 | mackes | Hmmmm. A Guess. Around $65,000 |
| 00:25.01 | mackes | I think |
| 00:25.06 | JT | bargain |
| 00:25.16 | JT | those dect bases are sounding temping |
| 00:25.17 | mackes | The system is for more the VoIP |
| 00:25.33 | drmessano | thepacmanfan: Every few months? That long? |
| 00:25.42 | drmessano | thepacmanfan: I was thinking 3 weeks |
| 00:25.44 | thepacmanfan | best case scenario. |
| 00:25.49 | drmessano | thepacmanfan: But hey, $50! |
| 00:25.57 | mackes | ok. never mind. I have to remember that when we have a discussion, it is always revolving around small scale. |
| 00:26.02 | thepacmanfan | i've had them last 3 months in PtP, but only 1 month on average for PtMP |
| 00:27.00 | drmessano | mackes: I know you think you're Mr Bigshit, and like to talk about how qualified you are, how big your installations are, etc.. but you're not impressing anyone with your little passive jabs about how no one else ever thinks as large scale as you do |
| 00:27.16 | JT | mackes: talking down to us really isn't going to do you any favours |
| 00:27.51 | *** join/#asterisk nighty^ (n=nighty@210.188.173.246) |
| 00:27.51 | JT | mackes: and the discussion has really confirmed what i already knew |
| 00:27.54 | jameswf-home | mackes: how many users do you put on a box |
| 00:28.00 | mackes | Look. I come online to talk about this stuff because i like it---- read over the thread and you will find that you two go after me every time I disagree. |
| 00:28.01 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 00:28.03 | JT | you need to spend big bucks to get wifi voip to work |
| 00:28.04 | drmessano | mackes: Many of us in here have done things that would blow your mind |
| 00:28.11 | JT | it might be the right solution for some |
| 00:28.14 | mackes | Great. |
| 00:28.16 | JT | but for most, it is not economical |
| 00:28.23 | Zorix | anyone else have any ideas how to fix the meetme voices for asterisk without zaptel hardware? |
| 00:28.25 | mackes | Then why are you so close minded? |
| 00:28.38 | JT | i'm not |
| 00:28.47 | JT | you just refuse to acknowledge some truths |
| 00:28.48 | jameswf-home | I made asterisk burn Linux CD's based on an IVR :) |
| 00:28.57 | mackes | " you just refuse to acknowledge some truths" |
| 00:29.01 | JT | Zorix: app_conference |
| 00:29.05 | jameswf-home | Zorix: ztdummy |
| 00:29.08 | mackes | ok, so, I am the issue here. |
| 00:29.18 | Zorix | ztdummy is loaded |
| 00:29.22 | Zorix | what is app_conference |
| 00:29.27 | JT | mackes: you need to spend a lot of money on infrastructure to make wifi voip work on any scale, you disagree with that? |
| 00:29.28 | drmessano | mackes: I'm far from close minded.. but I think your arguments are weak, you have little real proof to back up your statements, and when you dont have answers, you start trying to win out with your comments about how small we are thinking and how big your installs are.. it's pointless. |
| 00:29.37 | mackes | No, you do not. |
| 00:29.54 | mackes | I have one AP, and it covers my home--- two sip phones |
| 00:30.04 | JT | <PROTECTED> |
| 00:30.04 | JT | <PROTECTED> |
| 00:30.06 | jameswf-home | I have 1200 calls running through Asterisk on DSL using a trash *) |
| 00:30.11 | jameswf-home | *80 |
| 00:30.14 | JT | note the "on any scale" |
| 00:30.19 | JT | very selective reading |
| 00:30.42 | mackes | cool |
| 00:31.02 | drmessano | JT: My laptop makes calls using X-Lite over my WRT54G at home, so YOURE WRONG, BUDDY |
| 00:31.02 | mackes | ok guys, I can see this is your room, I am just a guest. |
| 00:31.04 | jameswf-home | I have windows vista on an old IBM XT |
| 00:31.21 | JT | mackes: so you don't even agree with what you've already told us? |
| 00:31.24 | mackes | What? Wrong about what? |
| 00:31.25 | JT | $65k isn't cheap |
| 00:31.33 | mackes | What? |
| 00:31.41 | JT | bah |
| 00:31.59 | mackes | Ok- Large Installs of SIP work |
| 00:32.07 | mackes | Small installs of WIFI Sip work |
| 00:32.10 | mackes | it all works |
| 00:32.17 | mackes | I think SIP WiFi works |
| 00:32.18 | Zorix | has ztdummy been purposely degraded so that people will buy digium hardware? |
| 00:32.22 | JT | if you spend lots of money |
| 00:32.23 | mackes | Those are my thoughts |
| 00:32.25 | mackes | no |
| 00:32.32 | JT | and yes, your retail installs are for large areas |
| 00:32.34 | mackes | You dont have to spend lots of money |
| 00:32.39 | mackes | yes |
| 00:32.40 | JT | they do not have large amount of users |
| 00:32.54 | mackes | I didnt say that |
| 00:33.04 | jameswf-home | Zorix you could buy a $10 clone card as a timing source |
| 00:33.09 | JT | so it's not a large install from a head count perspective |
| 00:33.16 | JT | Zorix: yes |
| 00:33.26 | drmessano | $65000 per 6 users |
| 00:33.27 | Zorix | jameswf-home yes but why, a reinstall of asterisk is all that broke it |
| 00:33.30 | thepacmanfan | do i need to modprobe zaptel after every reboot? |
| 00:33.35 | JT | there is no reason conferencing can possibly need zaptel timing |
| 00:33.35 | drmessano | Thats an $11000 wal mart phone |
| 00:33.39 | JT | it's just a legacy thing |
| 00:33.57 | mackes | The WiFi network support several hundred hand held Windows CE devices that are POS terminals as well. |
| 00:34.03 | mackes | And 80 laptop users |
| 00:34.14 | mackes | And a few other devices. |
| 00:34.26 | mackes | It is a busy WiFi network |
| 00:34.33 | mackes | The SIP phones are just a part |
| 00:34.43 | Zorix | interesting |
| 00:34.43 | mackes | And we only need so many AP's for coverage |
| 00:34.48 | Zorix | i did rmmod ztdummy |
| 00:35.06 | Zorix | and now i get a voice saying that is not a valid conference number when dialing my conference number |
| 00:35.13 | Zorix | but its the first voice that sounds good |
| 00:35.28 | JT | Zorix: i did mention before that app_meetme is zaptel conferencing |
| 00:35.35 | JT | no zaptel, no work :P |
| 00:35.51 | Zorix | right but the voice sounded correct |
| 00:35.51 | thepacmanfan | mackes, assuming 15 users per location that's nearly $5000 per user |
| 00:36.13 | JT | it's Playback |
| 00:36.23 | JT | it doesn't use zaptel |
| 00:36.27 | JT | of course it sounds good |
| 00:36.36 | jameswf-home | has been playing with shc... verry interesting |
| 00:36.37 | mackes | Over what period of time? |
| 00:37.00 | jeffspeff | if i have exten => *9,1,Dial(${jeffclay}/2705559026,30,mwW) and then from one cell phone i call into the system and go to exten *9 and it dials my cell phone... shouldn't at least one of the phones be able to initiate one-touch recording? |
| 00:37.16 | mackes | ROI is a tricky calculation |
| 00:37.57 | JT | if i don't want 80 laptop users or any pos terminal users, the ROI calculation would come back quite poor |
| 00:38.03 | Zorix | jameswf-home, where can i get that $10 card |
| 00:38.11 | jameswf-home | jeffspeff: Set(DYNAMIC_FEATURES=automon) |
| 00:38.16 | jameswf-home | Zorix: Ebay |
| 00:38.22 | Zorix | what is it called |
| 00:38.29 | jameswf-home | google x100p |
| 00:38.29 | JT | X100P |
| 00:38.33 | Zorix | ok thanks |
| 00:38.43 | JT | lots of fake x100p units on ebay |
| 00:38.47 | JT | Zorix: why not try app_conference instead? |
| 00:39.05 | Zorix | because im not looking to break asteriskgui |
| 00:39.09 | JT | lol |
| 00:39.17 | mackes | This is hard to discuss via chat- However, would you agree that many businesses are using WiFi Voip phones? Like Hospitals and Airports, and the government? |
| 00:39.18 | Zorix | trying to make this idiot proof |
| 00:39.30 | jameswf-home | maclno |
| 00:39.43 | jameswf-home | * mackes NO |
| 00:39.44 | JT | all those people have a lot of cash, mackes |
| 00:39.55 | drmessano | I would say no |
| 00:39.58 | jameswf-home | Zorix: they will just build a better idiot |
| 00:40.06 | Zorix | where can i get app_conference |
| 00:40.18 | jameswf-home | Deffinately not in hospitals to critical of an app |
| 00:40.20 | JT | and it is still not relevant, the fact that wifi voip phones are being user was *never* under dispute |
| 00:40.29 | JT | s/user/used/ |
| 00:40.58 | jameswf-home | I know nursing homes using wifi pdas for paging |
| 00:41.11 | drmessano | The dispute was whether or not they suck |
| 00:41.28 | mackes | Hmmm. Have you heard of Spectralink? |
| 00:41.31 | mackes | http://www.spectralink.com/home |
| 00:41.43 | JT | sounds like they stole the name from Motorola Spectra :P |
| 00:42.08 | jameswf-home | JT was on ignore? wonder why... I dont usualy ignore people |
| 00:42.08 | ManxPower | Spectra has the most expensive cordless phone I have *ever* seen. |
| 00:42.17 | mackes | All the Supermarkets in my area, and Kalida Heath care uses spectralink WiFi phones |
| 00:42.22 | *** join/#asterisk Robba (n=rob@203.56.181.15) |
| 00:42.23 | mackes | So? What is your point? |
| 00:42.46 | JT | if you can't see the point by now, you never will |
| 00:42.51 | JT | wifi is not a be all and end all solution |
| 00:43.02 | Robba | Hi guys |
| 00:43.05 | mackes | oh no, I see |
| 00:43.06 | jameswf-home | Kalida sounds like something you catch when you get drunk in thailand and go home with a local |
| 00:43.12 | JT | it suits certain requirements and often requires a large investment |
| 00:43.19 | Robba | does anyone know if its possible to change DTMF timing in * |
| 00:43.30 | jeffspeff | jameswf-home, I set the dynamic_features=automon in my globals in extensions.conf... i used one cell phone, called in, went to *9, it dialed my cell phone... the call was connected and, when either phone types in *1 (for one-touch recording) i get the following in the console... " -- Packet2Packet bridging SIP/jeffclay-09d43ef0 and SIP/teliax-09dc2d10" |
| 00:43.33 | drmessano | WOW |
| 00:43.38 | drmessano | all the news links on the site go nowhere |
| 00:43.44 | thepacmanfan | so what will cause a fast busy signal in asterisk? |
| 00:43.49 | mackes | Do any of you work for Digium? |
| 00:44.03 | JT | no, only people with operator status do |
| 00:44.16 | jameswf-home | jeffspeff: why dont you just record all calls on that context and be done with it |
| 00:44.44 | jameswf-home | you wanna call our boss? |
| 00:44.52 | ManxPower | I do not work for Digium |
| 00:44.52 | jameswf-home | :) |
| 00:45.07 | jameswf-home | works for a Digium competitor |
| 00:45.14 | drmessano | http://www.spectralink.com/products/index.jsp |
| 00:45.18 | ManxPower | gasps |
| 00:45.23 | jameswf-home | :) |
| 00:45.23 | drmessano | Four of the Six listed are not even Wifi |
| 00:45.48 | Zorix | JT, what zaptel hardware module does x100p use? |
| 00:45.52 | Robba | ok i'll take that as a no |
| 00:45.58 | ManxPower | IIRC Spectralink are $300-$500 each |
| 00:46.05 | Zorix | wcfxo maybe? |
| 00:46.09 | ManxPower | Zorix: Ambien |
| 00:46.25 | Zorix | isnt that sleep medication? |
| 00:46.31 | mackes | So, are Linksys, Hitachi, Cisco, Spectralink (Polycom), and Nokia all wrong, and you few are correct. |
| 00:46.38 | ManxPower | Oh, you mean what kernel module. Each card and the correct modules is listed in the Zaptel readme |
| 00:46.59 | Zorix | ok |
| 00:47.00 | JT | mackes: that was never the debate. keep it professional and on topic. it's not about "with us or against us" thinking. |
| 00:47.02 | ManxPower | mackes: Yeah, but the ones I looked at are just 900Mgz, 2.4Hgz cordless phones -- still 300 - 500 range |
| 00:47.36 | JT | manufacturers will always chase a dollar |
| 00:47.46 | drmessano | WIP300 <-- Rest my case |
| 00:47.47 | JT | and there's lots of dollars to chase in wifi voip |
| 00:47.52 | jameswf-home | thinks asterisk is an expensive hobby and cheap folks dont get far |
| 00:48.00 | jeffspeff | jameswf-home, I don't want to record every call... the *9 extension is for when a client calls in to the system when we're closed and needs emergency assistance... they dial *9 and it goes to my cell phone... I would like the option to record those calls if i want... one-touch recording doesn't seem to work when i do that... but, the recording works in every other aspect of the dial plan, even when i call *9 from one of the sip regi |
| 00:48.00 | jeffspeff | stered phones |
| 00:48.20 | jameswf-home | ~newbjuice |
| 00:48.21 | jbot | DUDE your spilling your red newb juice on the white carpet and we just had it cleaned... |
| 00:48.21 | drmessano | Grandstream sucks, and somehow they manage to make a few good dollars |
| 00:48.22 | ManxPower | jeffspeff: my guess is you really have a DTMF problem |
| 00:48.22 | JT | jameswf-home: i agree |
| 00:49.00 | jameswf-home | jeffspeff: add the record prior to the dial that calls your phone |
| 00:49.05 | JT | jeffspeff: you can still put MixMonitor in the dialplan for just that |
| 00:49.38 | jeffspeff | jameswf-home, how do i do that? |
| 00:49.44 | Robba | has anyone else had issues with asterisk and external IVR systems? |
| 00:49.55 | jameswf-home | jeffspeff: did you not write your dialplan? |
| 00:49.59 | jeffspeff | yes |
| 00:50.20 | jeffspeff | jameswf-home, but i don't know how to set it to record all calls like that |
| 00:50.25 | jameswf-home | http://www.voip-info.org/wiki/index.php?page=Asterisk+record+calls |
| 00:50.34 | ManxPower | Robba: only the standard / classic issues with DTMF and T/t/W/w dial opts |
| 00:50.55 | ManxPower | as well as audio gains and dtmfmodes |
| 00:50.58 | ManxPower | ~ask |
| 00:50.59 | jbot | extra, extra, read all about it, ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 00:51.49 | *** join/#asterisk PepOSX (n=angeldav@200.93.28.168) |
| 00:53.48 | Zorix | ok i think i know how i can fix this problem with the meetme voices, the ztdummy is using HRtimer instead of the UHCI usb which i think I was using last time, is there a way to change this without a recompile? |
| 00:54.33 | Robba | Manx: the issue seems to be when calling other systems with IVR eg. Banks, Carriers... when they ask for the key presses they contiually get rejected. |
| 00:55.03 | ManxPower | what are you dialing from? |
| 00:55.24 | Robba | asterisk with a TE122P card |
| 00:55.45 | ManxPower | play around with txgain for the zap channels. What phone are you using? |
| 00:55.54 | Robba | linksys SPA942 |
| 00:56.04 | JT | what codec is the phone using, with what dtmfmode? |
| 00:56.21 | Robba | RFC DMTF MODE and G.711U |
| 00:56.28 | ManxPower | the phone should be set for DTMF AVT, Asterisk should be set for dtmfmodr=rfc2833 |
| 00:58.25 | ManxPower | but if you have no other DTMF issues (no problems using an internal IVR, no voicemail, etc problems, then check your txgains |
| 00:58.33 | *** join/#asterisk s0lid (n=s0lid@210.213.198.219) |
| 01:01.25 | Robba | there is a section for the phone config "DTMF Playback Level" it is currently set to -16 |
| 01:01.35 | Robba | would this have anything to do with it? |
| 01:01.39 | ManxPower | Robba: that does not apply in this case. |
| 01:01.47 | Robba | oh ok thanks |
| 01:01.57 | ManxPower | it would either apply to playing dtmf over the ear piece or for inband DTMF. |
| 01:02.32 | Robba | so what level would be suggested? |
| 01:02.33 | ManxPower | Robba: also look in zapata.conf for the dtmf tone length. set it to betweem 300 and 500. You will have to unload/load chan_zap.so or restart asterisk, a reload won't do it. |
| 01:03.12 | ManxPower | rememeber, asterisk regenerates DTMF for sending out the zap port. |
| 01:04.03 | Robba | what string should i put in the zapata.conf to increase DTMF length i found a few command but it didn't seem to make any difference? |
| 01:10.27 | jameswf-home | didigoogleit=0 |
| 01:11.59 | Robba | sorry james but you aren't being much help |
| 01:12.28 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-ed9817fdf56e7c95) |
| 01:12.29 | jameswf-home | ~fire |
| 01:12.30 | jbot | Bender : Light a fire for a man and he's warm for a night. Light a man on fire and he's warm for the rest of his life... |
| 01:13.06 | dFence | <rant>perl annoys the crap out of his royal highness</rant> |
| 01:13.35 | jameswf-home | dFence: you could do it in php |
| 01:14.00 | Robba | for the txgain what would be an acceptable increase? |
| 01:14.11 | dFence | jameswf-home: i'm gonna tell the billing-app that explicitly requires perl and certain modules to go screw themselves and go php. |
| 01:14.30 | jameswf-home | i usualy increase gains 2 points at a time |
| 01:15.26 | Zorix | how can i change ztdummy source from hrtimer to rtc? |
| 01:15.48 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-f06c34b2016bc9ea) |
| 01:15.50 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 01:15.51 | Robba | thanks james i will give that a try |
| 01:15.51 | jbot | no problem, Robba |
| 01:15.57 | dFence | hm.. in my current mood i shouldn't be talking to other humanoid beings... night all! |
| 01:16.22 | JT | jbot seems to thinks it is being thanked if anyone starting with j is thanked |
| 01:16.23 | jbot | JT: what are you talking about? |
| 01:16.26 | JT | jbot: you |
| 01:16.26 | jbot | it has been said that jbot is a hack!, or known to have only said one useful thing. a tool, or dating the fembots, or [TK]D-Fender's b*tch, or suck, or a pain in the ass |
| 01:16.35 | *** join/#asterisk salzh (n=salzh@116.233.145.18) |
| 01:18.19 | jameswf-home | I should rewrite ztdummy |
| 01:18.30 | jameswf-home | just for the hell of it |
| 01:19.07 | rob0 | kram: thanks |
| 01:19.19 | jameswf-home | ok too zelous maybe patch it a bit |
| 01:34.06 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 01:57.02 | *** join/#asterisk uluatu (n=deg@189.58.13.91.adsl.gvt.net.br) |
| 01:59.31 | mackes | ~newbjuice |
| 02:02.57 | mackes | Hmmmm |
| 02:03.01 | mackes | Slowed Down |
| 02:04.47 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:14.20 | *** join/#asterisk V-chris (i=chris@asterisk.voice.li) |
| 02:17.13 | secgod | how do you get a phone directory to show up on eyebeam/x-lite ? |
| 02:18.11 | mackes | I think you have to create it from scatch. |
| 02:18.18 | mackes | Asterisk doesnt supply it |
| 02:19.23 | secgod | so if i had to deploy 20 softphones and 20 VOIP phones I would need to visit 40 devices to get a directory on them and then have to visit all 40 each time i made a change? |
| 02:20.56 | mackes | Well, for Eyebeem, you could push updates to the software using a fileshare to the programs installed directory- depending on OS |
| 02:21.40 | mackes | The VoIP phone might allow for one central file to be updated, and pulled via TFTP (or FTP) during the phones reboot. |
| 02:22.14 | secgod | mackes, which VoIP phones support this? |
| 02:22.22 | mackes | Ummm Most |
| 02:22.35 | mackes | Aastra, Polycom |
| 02:23.00 | mackes | They have a directory that can be updated in one place, and then that file is pulled by each phone at reboot |
| 02:23.19 | mackes | On most hardphones that is standard |
| 02:23.25 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-6009ca7cdafdfac5) |
| 02:23.30 | JT | what sort of idiot deploys 20 voip phones without central provisioning? :P |
| 02:23.44 | secgod | mackes, what is the difference between Aastra / Polycom ? |
| 02:23.48 | mackes | JT man, that sounds somewhat harsh |
| 02:24.01 | mackes | A mater of opinion |
| 02:24.37 | JT | mackes: either you're paid by the hour and want to rack up charges for the client, or you're silly if you deploy 20 voip phones without central provisioning |
| 02:24.43 | JT | a matter of sense |
| 02:24.43 | mackes | Aastra has alot of options, and are easy to set up |
| 02:24.49 | JT | it's very easy to set up provisioning |
| 02:24.55 | mackes | I agree. Its just the way you say it |
| 02:25.17 | mackes | Polycoms are more Professional feeling. |
| 02:25.28 | JT | what was wrong with the way i said it? |
| 02:25.40 | mackes | It doest matter |
| 02:25.58 | secgod | I see most recommending the Polycoms as well. :) |
| 02:26.28 | mackes | Yep. Polycom are very nice. The IBM of the SPM phone world |
| 02:26.39 | mackes | SIP not SPM |
| 02:27.08 | mackes | Polycoms are abit harder to setup- and provision the first time |
| 02:27.20 | secgod | mackes, which t1/pri cards do you suggest ? |
| 02:27.53 | mackes | I have only purchased Digium so far, so thats all I know |
| 02:27.58 | secgod | if Polycoms are like the Cisco 7940/7960 it should be fine |
| 02:28.11 | mackes | They are- kinda of. |
| 02:28.35 | mackes | The Polycom phones use an XML provisioning file that is somewhat confusing the first time |
| 02:29.03 | secgod | mackes, i take it you built your own PBX then? Any docs or places that recommend how to build your PC like how much memory, raid, processor, etc ? |
| 02:29.16 | mackes | Cisco 7940/60 and Aastra use a flat file for provisioning, that is very clear, and strait forward |
| 02:29.44 | mackes | For Asterisk? How many extensions? How many calls will be happening at one time? |
| 02:30.05 | secgod | 25 extensions - 8 - 10 calls at once |
| 02:30.22 | mackes | Ok, A inexpensive machine will do. |
| 02:30.26 | JT | secgod: make sure you get a card with hardware echo cancellation |
| 02:30.28 | mackes | A newer desktop machine |
| 02:30.39 | mackes | Yep, echo Cancellation for your PRI card |
| 02:31.04 | secgod | i am most likely going to use digium card so i assume this is standard right?> |
| 02:31.12 | mackes | A New dell desktop, with a P4, or Dual Core, and 512, to a 1GB of RAM will be more then enough |
| 02:31.26 | mackes | You are always safe with Digium |
| 02:31.41 | JT | lol |
| 02:31.49 | mackes | hear we go |
| 02:32.07 | mackes | Mr JT, what do you think> |
| 02:32.16 | JT | secgod: it's better to get a second hand server with redundant everything than a new desktop with redundant nothing |
| 02:32.17 | secgod | i see a bunch of knock offs on ebay but i can't risk it.. |
| 02:32.28 | mackes | Yep. |
| 02:32.29 | JT | secgod: sangomas are pretty decent too |
| 02:32.30 | mackes | I agree. |
| 02:32.32 | JT | they're not knock offs |
| 02:33.08 | mackes | I havent used them, but I have heard good things about sangoma |
| 02:33.27 | JT | they now have a lifetime warranty |
| 02:33.32 | JT | so imho it's a no brainer |
| 02:33.49 | mackes | If you can afford a new machine with a RAID, get it |
| 02:34.04 | secgod | ya definitely raid on this one |
| 02:34.09 | mackes | Many machines have SATA rad on the main board |
| 02:34.11 | secgod | so sangoma is good then? |
| 02:34.43 | mackes | I have not used them. When my company moved to Asterisk, I stuck with Diguim |
| 02:34.50 | mackes | Thats me |
| 02:35.13 | JT | we've replaced a lot of our customers digium cards with sangoma and the customers are now happy |
| 02:35.13 | *** join/#asterisk hardwire (n=hardwire@rdbk-17128.mtaonline.net) |
| 02:35.23 | mackes | And buying Digium supports Asterisk and Digium |
| 02:35.31 | JT | digium cards are often fine, but they just hate a lot of chipsets |
| 02:35.36 | mackes | My cards have never needed replacement |
| 02:35.38 | JT | and have poor zttest scores |
| 02:35.46 | JT | which result in dropouts |
| 02:35.49 | JT | and noise |
| 02:35.50 | secgod | JT, where can i get the sangoma cards ? |
| 02:35.58 | JT | telephonydepot.com |
| 02:36.02 | file | current generation cards should be fine, if not Digium wants to know so they can be |
| 02:36.10 | hardwire | is loading wcopenpci (which detects then fails to configure a TDM400 board before wctdm doing anything sucky to the init of the board? |
| 02:36.44 | JT | i've found there was no difference with latest digium cards, they just don't like some pcs |
| 02:37.15 | mackes | So, |
| 02:37.25 | secgod | JT, which model sangoma does T1/PRI ? |
| 02:37.26 | mackes | Does that help Secgod? |
| 02:37.33 | secgod | i see T1/E1 |
| 02:37.38 | JT | secgod: same thing |
| 02:37.41 | mackes | That is PRO |
| 02:37.44 | mackes | PRI |
| 02:37.48 | secgod | ok |
| 02:38.02 | mackes | PRI is a channelized T1 |
| 02:38.49 | secgod | ok. so just pick the card with the right port density is it then? |
| 02:38.59 | *** join/#asterisk hauptmech (n=th@ip-118-90-96-59.xdsl.xnet.co.nz) |
| 02:39.33 | mackes | Right- Do you need more then one PRI? |
| 02:39.39 | secgod | no just 1 |
| 02:39.53 | secgod | but i am thinking of looking at the dual card just to be safe |
| 02:39.53 | hardwire | you should buy 2 port cards |
| 02:39.59 | secgod | hehehe yep :) |
| 02:40.03 | hardwire | I constantly find myself 1 port shy. |
| 02:40.13 | hardwire | esp when I want to do testing. |
| 02:42.56 | secgod | thanks for the tips.. much appreciated... time to do more reading and research |
| 02:42.56 | jbot | secgod: de rien |
| 02:43.53 | hauptmech | newb here. does "--SIP/bla.net-083278 is circuit busy" mean I really connected to that endpoint? |
| 02:44.14 | secgod | wow.. glad i read the specs. the a102d has the echo canceller and the non D doesn't .. thanks |
| 02:44.58 | hauptmech | I suspect my routing but if it's connecting then I don't need to worry about it... |
| 02:55.15 | JT | mackes: PRI is a voice T1 or E1 running a D channel with PRI signalling |
| 02:55.34 | JT | most people in the telecom industry refer to a "channelized" T1 as a T1 running RBS voice |
| 02:55.52 | mackes | Hmmmmm. ok |
| 02:55.54 | mackes | Thanks |
| 02:55.54 | jbot | mackes: pas de quoi |
| 02:56.36 | mackes | Is a PRI a T1? |
| 02:56.46 | JT | a PRI runs over a T1 (or an E1) |
| 02:57.16 | mackes | so a PRI is a T1, or runs ontop of a T1? |
| 02:57.26 | JT | runs on top |
| 02:57.34 | JT | a T1 may or not have a PRI on it |
| 02:57.51 | JT | a PRI is really primary rate isdn, so has a d channel for signalling |
| 02:58.00 | mackes | So, a PRI could run on something other then a T1? |
| 02:58.15 | JT | yeah, here they run on E1s |
| 02:58.22 | JT | 30B + 1D |
| 02:58.28 | JT | instead of 23B + 1D |
| 02:58.45 | mackes | What if you have a Spilt- Half Data and half PRI? |
| 02:58.59 | JT | that's possible too |
| 02:59.11 | JT | just some timeslots are not used for voice then |
| 02:59.42 | mackes | I always thought a PRI/ T1 was the same thing- it just mattered if you were running voice channels or Data channels |
| 02:59.48 | mackes | All Data- T1 |
| 02:59.53 | *** join/#asterisk uluatu (n=deg@189.58.13.91.adsl.gvt.net.br) |
| 02:59.55 | mackes | All Voice PRI |
| 03:00.59 | JT | voice is not always PRI |
| 03:01.18 | mackes | I meant channelized voice |
| 03:01.20 | mackes | sorry |
| 03:01.27 | mackes | Well. That is interesting. |
| 03:01.57 | JT | in the US they also have channelised T1s for voice using Robbed Bit Signalling (Channel Associated Signalling) |
| 03:02.02 | JT | 24 voice channels |
| 03:02.26 | mackes | Could I use a Digium T1 card and connect a Linux box straight to a data T1 without a router? |
| 03:02.47 | JT | i believe so |
| 03:02.57 | mackes | neat |
| 03:03.22 | JT | as long as you have a CSU/DSU / Smartjack / SHDSL modem |
| 03:03.34 | mackes | Smartjacks |
| 03:03.56 | mackes | What do you do JT? |
| 03:04.02 | mackes | for a living> |
| 03:04.34 | JT | do it infrastructure and asterisk systems as a day job |
| 03:04.51 | JT | on the side i'm starting a voip and online faxing provider |
| 03:04.57 | JT | as well as co-location |
| 03:05.06 | mackes | One company? or are you a consultant? |
| 03:05.30 | JT | my day job i'm just an employee, my night job, that's my company... |
| 03:07.46 | *** join/#asterisk codefreeze-lap (n=murf@216.166.159.235) |
| 03:10.52 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 03:12.16 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-7d5b25a7d08e6b97) |
| 03:13.52 | *** join/#asterisk SanityIO (n=SanityIO@77.242.106.224) |
| 03:16.08 | jeffspeff | What am I missing here? I'm making a monitor macro so that i can define the file name when it's saved... currently my file name is defined as ---> exten => s,1,Set(CALLFILENAME=${CALLERID(num)}-${EXTEN}-${STRFTIME(,/usr/share/zoneinfo/America/Chicago,)}) and when I call extension 1001 from exten 1000 and start the macro, the resulting file name is ---> 1001-s-Sun May 18 22:12:10 2008.wav ... I see that it's getting the "s" |
| 03:16.08 | jeffspeff | from the exten=s part of the macro, but how do i get the file name to show the exten that I'm dialing from? |
| 03:16.54 | jeffspeff | the result i'm looking for is something like ---> 1001-1000-Sun May 18 22:12:10 2008.wav ..... instead of ----> 1001-s-Sun May 18 22:12:10 2008.wav |
| 03:17.39 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 03:18.05 | Strom_C | jeffspeff: in asterisk, extension != phone |
| 03:18.10 | *** join/#asterisk gardo (n=gardo@122.3.11.82) |
| 03:18.33 | Strom_C | also, you can use the MACRO_EXTEN variable to see which extension invoked the macro |
| 03:19.40 | jeffspeff | oh, so i could use ---> exten => s,1,Set(CALLFILENAME=${CALLERID(num)}-${MACRO-EXTEN}-${STRFTIME(,/usr/share/zoneinfo/America/Chicago,)}) ??? |
| 03:19.58 | *** join/#asterisk keith4_ (n=keith@216-164-160-36.c3-0.eas-ubr10.atw-eas.pa.static.cable.rcn.com) |
| 03:20.18 | keith4_ | ~itsp |
| 03:20.18 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 03:22.11 | jeffspeff | Strom_C, that resulted in 1001--Sun May 18 22:21:31 2008.wav as the file name |
| 03:22.27 | Strom_C | i think the variable name is MACRO_EXTEN, not MACRO-EXTEN |
| 03:22.50 | jeffspeff | ahh, my bad... let me try |
| 03:24.09 | jeffspeff | Strom_C, i changed the - to a _ and still had the same result of |
| 03:24.17 | jeffspeff | 1001--Sun May 18 22:21:31 2008.wav |
| 03:24.53 | Strom_C | that's within your macro? |
| 03:25.01 | Strom_C | that you're calling with the Macro() app? |
| 03:27.51 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 03:29.08 | jeffspeff | Strom_C, let me put the stuff in a pastebin for you, and i'll show you what i got... |
| 03:32.06 | jeffspeff | Strom_C, http://pastebin.ca/1022455 |
| 03:33.03 | *** join/#asterisk BeeBuu (n=beebuu@218.13.99.251) |
| 03:37.09 | BeeBuu | anyone know any good SIP component? |
| 03:37.26 | keith4_ | "component"? |
| 03:39.20 | jeffspeff | Strom_C, any ideas? |
| 03:40.59 | keith4_ | um |
| 03:42.34 | jameswf-home | Registrar Server is a sip component |
| 03:42.48 | keith4_ | if MACRO_EXTEN isn't defined, it doesn't think you're in a macro, no? |
| 03:42.50 | jameswf-home | UAS |
| 03:42.55 | jameswf-home | UAC |
| 03:47.36 | jeffspeff | keith4, did you see the pastebin? |
| 03:47.51 | keith4_ | yes |
| 03:48.15 | keith4_ | can you connect to the asterisk console, crank up verbosity, and pastebin that output? |
| 03:48.40 | jeffspeff | asterisk -rvvvvvvv good enough? |
| 03:49.50 | keith4_ | it's a start |
| 03:50.57 | jeffspeff | keith4, http://pastebin.ca/1022464 |
| 03:52.05 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:52.11 | jeffspeff | keith4_, http://pastebin.ca/1022464 |
| 03:52.27 | [TK]D-Fender | jeffspeff, thats not going to work because when Macro is called its not ON an exten. |
| 03:52.38 | [TK]D-Fender | jeffspeff, its being generated out of thin air |
| 03:53.18 | [TK]D-Fender | jeffspeff, see if you can use an inherited variable... I'm not sure how it spawns the channel that gets run in, but that might pan out better |
| 03:53.20 | keith4_ | explains why MACRO_EXTEN is empty |
| 03:53.41 | jeffspeff | <PROTECTED> |
| 03:54.48 | [TK]D-Fender | jeffspeff, I jsut told you what to try for this. Get to it. |
| 03:55.06 | keith4_ | what about using DIALEDPEERNAME or something similar? |
| 03:55.29 | keith4_ | there must be something predefined that would be appropriate |
| 03:55.39 | JT | why can't you just pass ${EXTEN} into the macro? |
| 03:55.58 | jeffspeff | ${EXTEN} returns with a value of "s" |
| 03:56.18 | JT | i said pass it in |
| 03:56.30 | JT | when you call the macro, pass ${EXTEN} as a macro var |
| 03:56.57 | jeffspeff | i'm confused... :s |
| 03:57.07 | JT | read up on Macro() |
| 03:57.22 | jeffspeff | the macro is being called from the application map in features.conf |
| 03:57.52 | jeffspeff | would it be better to call the macro from within extensions.conf |
| 03:57.54 | jeffspeff | ? |
| 03:58.02 | keith4_ | try inheritance |
| 03:58.12 | hardwire | hmmm /me wonders how to get this gxp-2000 off of the 1.0.1.8 firmware |
| 03:58.16 | hardwire | tftp isn't doing the trick |
| 04:10.14 | hardwire | ah |
| 04:15.07 | Zorix | i cant believe i have to buy hardware to get audio to work correctly...stupid crippled ztdummy |
| 04:15.38 | Zorix | shouldn't have upgraded |
| 04:16.50 | hardwire | I think you are silly |
| 04:16.55 | [TK]D-Fender | Zorix, Go downgrade then. |
| 04:17.16 | Zorix | zttest shows accuracy is -200% |
| 04:17.26 | Zorix | i cant find my old cd or i would |
| 04:18.31 | hardwire | Zorix: sounds like your the right guy for the job. |
| 04:18.50 | hardwire | sry.. you're |
| 04:18.50 | hardwire | :) |
| 04:18.54 | Zorix | been fightin this thing for 4 hours |
| 04:19.35 | hardwire | I'm slowly upgrading some grandstreams |
| 04:19.43 | hardwire | I'm getting all excited when it works |
| 04:19.57 | [TK]D-Fender | Zorix, ... CD? |
| 04:20.05 | hardwire | [TK]D-Fender: I know, right? |
| 04:20.14 | hardwire | Maybe it's asterisk enterprise? |
| 04:20.17 | Zorix | [TK]D-Fender, cd-r |
| 04:20.25 | [TK]D-Fender | Zorix, what CD? |
| 04:20.29 | hardwire | haha |
| 04:20.32 | hardwire | sigs |
| 04:20.34 | hardwire | hs |
| 04:20.35 | Zorix | old version of asterisknow |
| 04:20.53 | hardwire | it would suck having to re-download it? |
| 04:20.57 | [TK]D-Fender | Zorix, Go find out what version it was using and jsut re DL it. |
| 04:21.03 | Zorix | yes it would not knowing which one it was |
| 04:21.20 | [TK]D-Fender | Zorix, Don't know what version it was? |
| 04:21.30 | Zorix | it was over a year ago i installed it |
| 04:21.53 | [TK]D-Fender | Zorix, Hoefully this taught you a few lessons |
| 04:21.55 | hardwire | Zorix: more to the point, what hardware did you buy and what machine is a newer AsteriskNOW running on? |
| 04:22.08 | Zorix | i didnt buy any hardware |
| 04:22.23 | Zorix | yea never trust asterisk upgrades thats what its taught me |
| 04:23.38 | hardwire | I'm thinking you need to hit your head on a few more toilets |
| 04:23.46 | [TK]D-Fender | Zorix, Wrong lesson. |
| 04:23.56 | hardwire | test first |
| 04:24.00 | hardwire | be prepared |
| 04:24.05 | hardwire | always have your towel handy. |
| 04:24.25 | [TK]D-Fender | Zorix, Do stuff blind not having a clue what you've even got at the start and you get what you deserve. |
| 04:24.32 | Zorix | i asked questions here and the only answer i got was to buy zaptel hardware for the timer |
| 04:24.41 | Zorix | its not a production system |
| 04:24.43 | hardwire | Zorix: did you learn your lesson then? |
| 04:24.47 | [TK]D-Fender | Zorix, And you couldn't think to downgrade? |
| 04:24.50 | hardwire | you didn't buy the hardware. |
| 04:25.08 | hardwire | and you appear frustrated with truthful answers. |
| 04:25.16 | Zorix | i didnt have to have hardware for the older install |
| 04:25.39 | hardwire | what machine was this running on? |
| 04:25.44 | Zorix | same one it is now |
| 04:25.59 | hardwire | should I ask again? |
| 04:26.00 | Zorix | i believe it was using the usb uhci timer |
| 04:26.11 | Zorix | its some old hp pavilion i dont know exact model |
| 04:26.28 | hardwire | no usb 2.0? |
| 04:26.32 | Zorix | no |
| 04:28.17 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 04:28.17 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.4.20-rc3, 1.6.0-beta9 (2008/05/14) Asterisk 1.4.19.2 (2008/05/13), *-Addons 1.4.6 (2008/02/21) 1.6.0-beta3 (2008/04/02), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 04:28.24 | Zorix | i think thats some sort of kernel timer |
| 04:28.25 | hardwire | [TK]D-Fender: whats newer asterisk now implementations use? |
| 04:28.33 | [TK]D-Fender | Zorix, Things that can screw you up is kernel timer mismatch. |
| 04:28.58 | [TK]D-Fender | hardwire, all but certainly 2.6 |
| 04:29.03 | Zorix | wonder what the kernel timer is at |
| 04:29.08 | Zorix | supposed to be 1000 i think |
| 04:29.25 | [TK]D-Fender | Zorix, indeed it should be |
| 04:29.32 | Zorix | 2.6.22.13-0.1 |
| 04:29.43 | hardwire | Zorix: asterisk all around is failing? |
| 04:29.45 | Zorix | but i think i have to recompile to change it.. i dont know enough about 2.6 kernel |
| 04:29.50 | hardwire | or is it just meetme conferences? |
| 04:29.55 | Zorix | nope just meetme |
| 04:29.59 | Zorix | conference |
| 04:30.03 | hardwire | do you ever use it? |
| 04:30.05 | Zorix | i can do other voice calls just fine |
| 04:30.21 | Zorix | not often but knowing there's a big problem isnt good |
| 04:30.28 | Zorix | and i heard it affects iax trunking |
| 04:30.40 | hardwire | do you trunk IAX? |
| 04:30.49 | Zorix | thats one thing i will be using |
| 04:30.58 | hardwire | can you do me a huge favor? |
| 04:31.14 | hardwire | install ubuntu 8.04 (Alternate CD) then apt-get install asterisk |
| 04:31.19 | hardwire | see if you still have issues |
| 04:31.25 | Zorix | hehe |
| 04:31.39 | hardwire | got nothing to lose, right? |
| 04:31.44 | Zorix | i figured asterisknow would be better at it |
| 04:31.52 | Zorix | yea i do.. the hd is too small |
| 04:31.53 | hardwire | Zorix: better at what? |
| 04:31.57 | hardwire | packaging asterisk? |
| 04:32.01 | Zorix | better at running asterisk |
| 04:32.04 | Zorix | since its by the same people |
| 04:32.34 | hardwire | I'm not gonna say it would be, but I use ubuntu and the asterisk 1.4.x packages |
| 04:32.35 | Zorix | and not enough ram to run ubuntu |
| 04:32.38 | hardwire | works fine on most hardware |
| 04:32.54 | hardwire | Zorix: what makes you think you can't run ubuntu when you can run asterisknow? |
| 04:33.11 | Zorix | because asterisknow is minimal set of software to get it running |
| 04:33.19 | Zorix | i run ubuntu on my desktops |
| 04:33.21 | Zorix | so i like it |
| 04:33.31 | *** join/#asterisk Defraz (i=t0tal@69.92.19.83) |
| 04:33.42 | hardwire | well, ever installed ubuntu in CLI mode? |
| 04:34.00 | hardwire | boot alternate cd - select cli as boot alias |
| 04:34.04 | Zorix | i have in vm yes |
| 04:34.15 | hardwire | and I bet it made your VM burst into flames, correct? |
| 04:34.32 | hardwire | rPath is a pretty big bird too. |
| 04:34.41 | hardwire | it's streamlined - sure |
| 04:34.47 | Zorix | actually it did, it was a beta release |
| 04:34.50 | hardwire | but ubuntu in "minimal" mode is pretty tiny |
| 04:35.05 | [TK]D-Fender | Zorix, You're running ZAPTEL in a VM?!?! LOL |
| 04:35.11 | Zorix | no im not |
| 04:35.16 | [TK]D-Fender | Zorix, I sure hope not |
| 04:35.18 | hardwire | [TK]D-Fender: I gave him more credit than that. |
| 04:35.31 | Zorix | thanks for butting into the conversation, injecting a comment without reading the context |
| 04:35.32 | hardwire | [TK]D-Fender: I used ztdummy in an openvz tho, yesterday, worked cherry |
| 04:35.36 | [TK]D-Fender | hardwire, I don't give credit. Debit or cash only ;) |
| 04:35.47 | hardwire | [TK]D-Fender: insert your pin here. |
| 04:36.09 | [TK]D-Fender | sticks a pin through the eye of his hardwire effigy... |
| 04:36.12 | Zorix | im lookin for a way to do zaprtc |
| 04:36.26 | jameswf-home | wow I just pulled up southpark s01e01 huge difference between no budget and big budget |
| 04:36.27 | hardwire | Zorix: does asterisknow even have the linux headers/gcc ? |
| 04:36.41 | hardwire | install ubuntu, test ztdummy |
| 04:36.42 | [TK]D-Fender | Zorix, So what is your * environment that you feel Ubuntu is too "heavy"? |
| 04:36.48 | Zorix | has no kernel sources thats for sure |
| 04:36.57 | Zorix | appears to have gcc |
| 04:36.59 | hardwire | "apt-get install module-assistant; m-a prepare; m-a a-i -t zaptel-source" |
| 04:37.27 | Zorix | [TK]D-Fender, 500mhz p3, 96mb ram 10gb hd |
| 04:37.40 | [TK]D-Fender | Zorix, EEK. |
| 04:37.53 | jameswf-home | careful debians zaptel is all effed up Ubuntus may be too |
| 04:37.58 | Zorix | worked just fine on older asterisk |
| 04:38.12 | *** join/#asterisk rajiv (n=rajiv@gentoo/developer/rajiv) |
| 04:38.37 | *** join/#asterisk mkillebrew (n=mkillebr@50ae.net) |
| 04:39.16 | jameswf-home | you really should build asterisk/zaptel yourself god only knows what the packagers do |
| 04:39.19 | mkillebrew | is there a way to get asterisk to use the callerid=nxxnxxxx in sip.conf on set(callerid(num)) on outgoing in extensions.conf? |
| 04:39.51 | jameswf-home | if it worked fine on the older asterisk wtf did you touch it |
| 04:40.44 | [TK]D-Fender | mkillebrew, Sorry that came out a bit disjointed pastebin what you're doing now and tell us how its not performing as expected. |
| 04:40.46 | [TK]D-Fender | ~pb |
| 04:40.47 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 04:40.48 | [TK]D-Fender | ^^^^^^^^ |
| 04:41.00 | Zorix | jameswf-home im wishing i didnt right now but the older asteirsk was alpha or beta |
| 04:42.04 | mkillebrew | you mean it should work? maybe I'm just declaring the main number explicitly somewhere. |
| 04:42.31 | hardwire | jameswf-home: most of that is fixed now (eff-ed up zap) |
| 04:43.05 | [TK]D-Fender | mkillebrew, I mean you should show us what you're doing in detail (configs & CLI output) and show us where it goes wrong. |
| 04:58.32 | jeffspeff | ok, i found that if i use --- exten => s,1,Set(CALLFILENAME=${CALLERID(num)}-${DIALEDPEERNUMBER}-${STRFTIME(,/usr/share/zoneinfo/America/Chicago,)})--- then i get my desired result of --- 1000-1001-Sun May 18 23:48:37 2008.wav--- but, when i try to call an outbound line i get the following errors http://pastebin.ca/1022487 |
| 04:58.58 | mkillebrew | I just had it explicitly declared for some reason |
| 04:59.03 | *** join/#asterisk xenonex (n=xenonex@89.218.233.68) |
| 04:59.08 | mkillebrew | apparently it just works (tm) |
| 05:01.40 | [TK]D-Fender | jeffspeff, I don't believe you can call monitor in the MIDDLE of a call like that. |
| 05:04.37 | jeffspeff | [TK]D-Fender, ok, let me go back to square 1... I was using one-touch monitoring as defined in features.conf... I wanted to modify the way it was saving the files, so I thought the best way would be to write a monitor script. Obviously I'm going about this all wrong... what would you suggest? My desired filename format is something like Caller-Callee-Date.wave |
| 05:04.47 | jeffspeff | *.wav |
| 05:05.11 | [TK]D-Fender | jeffspeff, You should be able to set the file before you ever call dial. |
| 05:05.52 | jeffspeff | [TK]D-Fender, how's that? let me pastebin my extensions.conf |
| 05:07.27 | [TK]D-Fender | jeffspeff, You clearly didn't bother reading "channelvariables.txt" |
| 05:07.33 | [TK]D-Fender | jeffspeff, go read it. |
| 05:08.29 | jeffspeff | [TK]D-Fender, where is that file? |
| 05:08.56 | jeffspeff | [TK]D-Fender, here is my extensions.conf http://pastebin.ca/1022491 |
| 05:09.15 | [TK]D-Fender | jeffspeff, in the docs/ folder of your source tarball. |
| 05:17.28 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
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| 05:29.42 | jeffspeff | [TK]D-Fender, ok, i set MONITOR_FILENAME=(${CALLERID(num)}-${EXTEN}-${STRFTIME(,/usr/share/zoneinfo/America/Chicago,)}) in my globals in extensions.conf... before i did this, my filenames looked like auto-1210902844-SIP-1000-08913f08-918705550481.wav... now they look like auto-1211174398-1000-1001.wav |
| 05:30.05 | jeffspeff | why isn't it specifying the date, and why is it still doing the auto... thing? |
| 05:30.22 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.139) |
| 05:33.17 | [TK]D-Fender | jeffspeff, you can't set that in globals.. hte vars its based on are RUNTIME <- |
| 05:33.57 | [TK]D-Fender | jeffspeff, You are outsmarting yourself constantly and the failures predictable. |
| 05:35.12 | jeffspeff | SomethingISODD, do i have to do a exten=>s,1,Set(MONITOR_FILENAME=.........) inside every context that has a ,wW in a dial? |
| 05:35.17 | jeffspeff | *so, do i.... |
| 05:36.01 | hardwire | passes out, night all. |
| 05:36.32 | [TK]D-Fender | jeffspeff, read the big print. |
| 05:36.33 | *** join/#asterisk talntid (n=t@c-67-185-237-158.hsd1.wa.comcast.net) |
| 05:36.40 | jeffspeff | [TK]D-Fender, I appreciate your help as always, but I'm confused... the .txt file didn't say where that var is called or anything other than that the var existed and what it did. |
| 05:37.14 | [TK]D-Fender | jeffspeff, its all right htere in front of you. |
| 05:37.33 | jeffspeff | what big print? |
| 05:38.32 | [TK]D-Fender | jeffspeff, Yes you have to set this varible befoer your Dial, NO, you can't set it as a global. |
| 05:44.42 | *** join/#asterisk talntid (n=eric@c-67-185-237-158.hsd1.wa.comcast.net) |
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| 05:47.05 | jeffspeff | [TK]D-Fender, ok, here's what i did... I set the var in the extension right before dial (you can see it here from my extensions.conf http://pastebin.ca/1022503) When i dial ext 1001 the console shows that MONITOR_FILENAME is correctly set, but when it starts recording, it doesn't use the filename specified... see http://pastebin.ca/1022502 for the cli output |
| 05:47.35 | *** join/#asterisk Keltus (i=Keltus@about/cooking/nakedchef/beefstew/Keltus) |
| 05:47.44 | [TK]D-Fender | jeffspeff, because you aren't setting the right variable. |
| 05:48.09 | jeffspeff | i'm supposed to inlcude the $ and brackets aren't i? |
| 05:49.09 | [TK]D-Fender | jeffspeff, go read the docs again |
| 05:50.29 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 05:51.30 | *** join/#asterisk vortex` (n=vortex@202-136-108-213.static.adam.com.au) |
| 05:52.05 | vortex` | Just reading the asterisk book 'hello world' example at http://www.the-asterisk-book.com/unstable/minimale-telefonanlage.html and I have a question about a line in the config the book doesnt explain very well. |
| 05:52.33 | vortex` | the line refers to making calls to the PSTN from an Asterisk server & SIP phone |
| 05:52.50 | vortex` | the line is: exten => _0[1-9].,1,Dial(SIP/${EXTEN}@ext-sip-account) |
| 05:53.03 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 05:53.04 | vortex` | I'm not sure what the _0[1-9]. refers to as the first argument. |
| 05:53.20 | vortex` | is that a regex, saying any number 0 - 9 gets an outside line? |
| 05:53.34 | vortex` | (i understand the priority and function arguments, though) |
| 05:54.04 | [TK]D-Fender | vortex`, that is a pattern. when a call arrives into * from a device that is configured to use that context it will try to match the number dialed against that pattern. If it does, then it will process that exten accordingly. |
| 05:55.29 | [TK]D-Fender | vortex`, that particular pattern means starts with a "0", id followed by a digit "1-9", and then one or more characters of any kind |
| 05:56.11 | vortex` | [TK]D-Fender: thanks, yeah having read over the the section again it does actually mention it (but nothing about it being a pattern) :) |
| 05:58.23 | [TK]D-Fender | vortex`, that site does not appear to be a very complete way to learn * then. The dialplan is the most important part of *. If that guide doesn't explain how extens work then you should look elsewhere |
| 05:58.54 | vortex` | it's only the first example example aimed at a newbie, the rest of the site i assume goes into more detail. |
| 05:59.12 | vortex` | this first example will be good enough to get me started, however. |
| 05:59.21 | [TK]D-Fender | vortex`, here's hoping... |
| 05:59.24 | vortex` | :) |
| 05:59.26 | JT | ~thebook |
| 05:59.26 | 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 --- HTML at http://tfot.leifmadsen.com |
| 05:59.27 | *** join/#asterisk keulin (n=cray@80.15.251.6) |
| 05:59.47 | vortex` | JT: thanks; ooer, free PDF. i like the sound of that :) |
| 06:00.08 | JT | it's the bible |
| 06:00.35 | vortex` | as many oreilly books are on their specific subjects :) |
| 06:05.15 | jeffspeff | [TK]D-Fender, ok, i got it working much better now... I should have been setting the TOUCH_MONITOR var instead... but how can I get rid of the auto-.... that's still at the beginning of the files? |
| 06:05.30 | *** part/#asterisk vortex` (n=vortex@202-136-108-213.static.adam.com.au) |
| 06:05.48 | [TK]D-Fender | jeffspeff, Go try to do this properly and show me to full deal |
| 06:06.18 | jeffspeff | ok, i'll pastebin what i've set, and the cli ouput... just a sec |
| 06:10.00 | jeffspeff | [TK]D-Fender, here's the extensions.conf part... http://pastebin.ca/1022512 and here's the cli output http://pastebin.ca/1022514 and here is the actual file name auto-1211177265-1000-1001-Mon May 19 01:07:37 2008.wav |
| 06:11.53 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-d05e09fb3bbe3258) |
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| 06:12.21 | *** part/#asterisk jelly-bean (n=jelly-be@63-76-119-176.directcom.com) |
| 06:14.37 | [TK]D-Fender | jeffspeff, that appears fixes |
| 06:14.40 | [TK]D-Fender | jeffspeff, that appears fixesfixed* |
| 06:15.17 | jeffspeff | yes, but is there a way to get rid of the auto-1211177265- part? |
| 06:20.00 | [TK]D-Fender | jeffspeff, You are hard of hearing aren't you.... |
| 06:20.10 | [TK]D-Fender | jeffspeff, it appears to be FIXED <-------- |
| 06:20.56 | jeffspeff | oh, fixed as in can't get rid of it... i thought you were saying fixed as in i fixed my problem... |
| 06:21.16 | jeffspeff | thanks. :) |
| 06:22.21 | [TK]D-Fender | jeffspeff, the code the determines this is evident in res_features.c |
| 06:22.32 | [TK]D-Fender | jeffspeff, if you cared enough you could path this easily. |
| 06:22.35 | [TK]D-Fender | patch |
| 06:23.22 | jeffspeff | [TK]D-Fender, thanks, i'll take a look at that. |
| 06:23.37 | jeffspeff | once again, i've learned from you. :p |
| 06:24.52 | jeffspeff | [TK]D-Fender, I have to recompile after i change that right? |
| 06:26.09 | [TK]D-Fender | jeffspeff, unless the compile fairie will psychically know of your change and do it for you you mean? |
| 06:26.45 | jeffspeff | lol, whats the exten for the recompile fairy? i haven't seen that in any of the docs? is it built in? |
| 06:26.49 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 06:26.55 | jeffspeff | j/k |
| 06:28.04 | [TK]D-Fender | jeffspeff, Now what I might suggest is that its behaviour is by default illocical and worthy of being changed, thus you might gain some karma by posting a "fix" to it on Mantis. |
| 06:29.06 | jeffspeff | [TK]D-Fender, true; i'll see if i can find what to change, recompile, test, and then post the changed code. :) |
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| 06:46.51 | L|NUX | hello every one |
| 06:47.02 | jeffspeff | hello |
| 06:48.06 | L|NUX | hey jeffspeff |
| 06:48.08 | *** part/#asterisk Wi_Fi (n=OUT@cpe-76-169-21-21.socal.res.rr.com) |
| 06:48.17 | L|NUX | any one fimilar with DUNDi ? |
| 06:51.41 | jeffspeff | [TK]D-Fender, Give me your opinion on something. I've implemented a php page to review, play, and delete the recorded conversations.... I had another idea on somehow using a page that would allow a user to directly modify part of a series of dial commands within a certain context. my end result that i'm looking for is to type in a few numbers into the page and click save; and a extension dials those numbers and upon answering the u |
| 06:51.42 | jeffspeff | sers are transfered into the same conference... how would i best go about this? |
| 06:52.23 | jeffspeff | L|NUX: no experience with dundi |
| 06:53.21 | mvanbaak | L|NUX: just ask your question, maybe someone can help you |
| 06:54.08 | aiksa[LV] | jeffspeff: create AMI connection. Then issue Dial command |
| 06:54.41 | aiksa[LV] | where one part would be the location of the extension you would like to ring, like Local/15@internal/n |
| 06:54.54 | aiksa[LV] | and the other would be the location of the conference service |
| 06:54.54 | L|NUX | mvanbaak: okay i have setup dundi using http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP |
| 06:55.47 | jeffspeff | aiksa[LV], I'm not familiar with AMI, how involved is this idea of mine going to be? |
| 06:55.50 | L|NUX | but when i register on Box-A and calling Box-B like this 101@BoX-B it will say [May 19 01:44:43] NOTICE[11840]: chan_sip.c:13888 handle_request_invite: Failed to authenticate user "xxxxxxxxxxx" |
| 06:56.05 | aiksa[LV] | jeffspeff: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Originate |
| 06:56.27 | aiksa[LV] | what do you mean by involved (sorry i am not native speaker)? |
| 06:56.43 | L|NUX | my point is why call is not going to BoX-A |
| 06:57.15 | L|NUX | any one ? |
| 06:57.42 | jeffspeff | aiksa[LV], invovled, as in difficulty |
| 06:57.59 | aiksa[LV] | shouldnt be very difficult |
| 06:58.09 | jeffspeff | hmm.. ok |
| 06:58.13 | aiksa[LV] | AMI code of no more than some 20 line |
| 06:58.26 | [TK]D-Fender | jeffspeff, your description mixes too much about "how" you think you will go about doing what you really want to do, |
| 06:59.49 | jeffspeff | [TK]D-Fender, well, i would like it to work through a web interface... but have no idea exactly how to do it... aiksa[LV] recommended AMI; so i guess i'll read up on that. |
| 07:00.24 | aiksa[LV] | jeffspeff: you could also avoid AMI and just create dial spool files |
| 07:00.50 | [TK]D-Fender | jeffspeff, Youre descpriotion of what you want to DO was mangled in your description of HOW you though you should do it. Describe what you actaully want please and leave out the unneccesary "hows". |
| 07:00.51 | jeffspeff | aiksa[LV], what are dial spool files? never heard of those either |
| 07:01.04 | JT | .call files |
| 07:01.06 | aiksa[LV] | jeffspeff: take a look here: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out |
| 07:01.14 | JT | read sample.call in the source directory |
| 07:01.25 | aiksa[LV] | the page describes methods for automated dial out |
| 07:03.26 | jeffspeff | [TK]D-Fender, I want a webpage that a user can put phone numbers or extension into as well as desired conference room # and upon saving or whatever asterisk will dial the specified numbers, when the person on the other end asnwers, it will play a message like "please wait to enter conference" and then transfer all of those numbers to the same conference room. |
| 07:04.00 | [TK]D-Fender | jeffspeff, then read up on "call files" and "AMI originate" on the WIKI |
| 07:04.24 | mvanbaak | L|NUX: did you setup the sip.conf peer ? |
| 07:04.25 | jeffspeff | [TK]D-Fender, ok, thanks |
| 07:05.02 | JT | jeffspeff: read sample.call |
| 07:06.03 | *** join/#asterisk Maliuta (n=nikolai@kiev.lusan.id.au) |
| 07:06.04 | L|NUX | mvanbaak: i did |
| 07:06.37 | aiksa[LV] | jeffspeff: if you do it in pure "oldschool php" fashion. You might prefer call files better |
| 07:06.38 | *** join/#asterisk jelly-bean (i=user@63-76-119-176.directcom.com) |
| 07:07.15 | aiksa[LV] | as messing with AMI involves establishing sockets and operating in daemon mode |
| 07:07.51 | jelly-bean | how easy would it be to setup an asterisk box with a VoIP provider? what hardware would be necessary? i'm thinking a T1 with all 24 channels for data, a linux box to run asterisk, and a NIC + router |
| 07:07.58 | jelly-bean | am i right? |
| 07:08.37 | Strom_C | jelly-bean: or you could just put the T1 right into the asterisk box |
| 07:08.39 | aiksa[LV] | why would you need T1? |
| 07:08.53 | Strom_C | aiksa[LV]: symmetric bandwidth |
| 07:09.02 | aiksa[LV] | ok. |
| 07:09.52 | aiksa[LV] | Strom_C: here it would be cheaper to get an Optic rather than T1/E1 |
| 07:10.05 | Strom_C | well, congratulations? |
| 07:10.22 | JT | aiksa[LV]: wtf is an optic? |
| 07:10.27 | Strom_C | i think in English we call that "fiber" |
| 07:10.30 | aiksa[LV] | optical fiber |
| 07:10.40 | L|NUX | lol |
| 07:10.44 | JT | that's like saying "getting a copper" - meaningless |
| 07:10.45 | Strom_C | or "fibre" ;) |
| 07:10.51 | JT | you can run a LOT of stuff over fibre |
| 07:10.54 | JT | just like copper |
| 07:11.01 | L|NUX | i agree with JT |
| 07:11.09 | Strom_C | yeah -- what DSx level does the "optic" handle? |
| 07:11.18 | jeffspeff | aiksa[LV], JT, I've read the sample.call file, but i don't see anywhere to add the numbers that i want to dial |
| 07:11.50 | aiksa[LV] | JT, Strom_C point taken. I should go in the corner and be ashamed of myself |
| 07:12.17 | JT | aiksa[LV]: fibre is usually more expensive anyway, but for what it's worth, T1s and E1s can be provided over fibre too |
| 07:12.40 | JT | although most of the time they're provided over SHDSL over twisted pair copper |
| 07:12.52 | *** join/#asterisk cjk (n=cjk@vodsl-11071.vo.lu) |
| 07:13.17 | jelly-bean | Strom_C: "put the T1 right into the box"? how do you mean? I thought that is what i was saying :) |
| 07:13.27 | Strom_C | jelly-bean: put a T1 card in the machine |
| 07:13.45 | aiksa[LV] | JT, what I wanted to say, was that ifthe guy could get a provider to give him internet access over ordinary ethernet, he wouldnt need T1. |
| 07:14.11 | aiksa[LV] | provided that it is symetrical, stable and fast enough. |
| 07:14.17 | aiksa[LV] | or did i miss something? |
| 07:14.46 | aiksa[LV] | jeffspeff, there is Channel parameter in the call file |
| 07:14.47 | JT | most people aren't lucky enough to get cheap ethernet Internet to their premises |
| 07:15.15 | aiksa[LV] | JT: i must be thinking "locally". |
| 07:15.31 | jeffspeff | aiksa[LV], yes, but it said Only one channel name is permitted. |
| 07:15.32 | *** join/#asterisk pa (n=pa@unaffiliated/pa) |
| 07:15.43 | aiksa[LV] | jeffspeff: so create 10 or 20 files |
| 07:16.00 | aiksa[LV] | as many as the calls which has to be initiated. |
| 07:16.06 | aiksa[LV] | One file per one call |
| 07:16.07 | jeffspeff | aiksa[LV], didn't think about that |
| 07:17.43 | aiksa[LV] | JT as here I would pay around 900 EUR for E1 connection between two points. However guaranteed symetric connection to Internet with 8Mbits /8Mbits would go for 150 EUR |
| 07:19.06 | JT | aiksa[LV]: here a 2M/2M SHDSL connection would also be much cheaper than a point to point E1, even though an E1 is delivered over SHDSL |
| 07:19.25 | JT | T1 being a popular data connection method is really just a US/Canada thing these days I think |
| 07:19.52 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 07:20.09 | JT | aiksa[LV]: An an E1 connection to a telco here for PRI service is cheaper than 2M/2M SHDSL even... go figure :) |
| 07:20.36 | aiksa[LV] | JT not a case here |
| 07:20.55 | aiksa[LV] | I would have to pay EUR 300 just a monthly subscription fee to use E1 connection |
| 07:21.13 | aiksa[LV] | for PRI |
| 07:21.36 | jelly-bean | what is the speed of T1? or are there multiple speeds? Comcast is offering cable with 8mb down |
| 07:21.44 | jelly-bean | burstable up to 12mb down |
| 07:21.46 | aiksa[LV] | I guess thats why 90% of our pbxes end up with a number of BRI cards :P |
| 07:22.20 | Strom_C | jelly-bean: T1 is 1.544 megabits per second symmetric |
| 07:22.36 | JT | aiksa[LV]: here i can get a 10 channel PRI over E1 for the equivalent of 100EUR/mo |
| 07:22.47 | JT | jeffspeff: 1.54Mbit/s |
| 07:22.52 | JT | err |
| 07:22.56 | JT | jelly-bean: |
| 07:23.02 | jeffspeff | lol |
| 07:23.20 | aiksa[LV] | JT, nice. Where such paradise exists? |
| 07:23.29 | JT | we have ADSL2+ here, up to 24Mbit/s down, 1.2Mbit/s up |
| 07:23.31 | JT | Australia |
| 07:23.35 | aiksa[LV] | ok. |
| 07:24.01 | aiksa[LV] | JT and none of teh oprators here would ever agree to give you a part of full E1 timeframe count |
| 07:24.12 | aiksa[LV] | "the operators", sorry |
| 07:24.24 | JT | Internet transit is expensive here though |
| 07:24.26 | JT | aiksa[LV]: why not? |
| 07:24.48 | aiksa[LV] | JT, they say it is using up a "whole port for them" |
| 07:24.55 | JT | lol |
| 07:25.07 | JT | maybe if they have to buy a whole port from an upstream telco? |
| 07:25.31 | JT | i can get anything from ISDN 10 up to ISDN 30 |
| 07:25.34 | JT | depending on telco |
| 07:25.40 | aiksa[LV] | As in terms of their telco equipment. |
| 07:25.47 | JT | some telcos will allow an arbitary number of channels from 10 up |
| 07:25.52 | JT | others go in lots of 10 |
| 07:26.05 | aiksa[LV] | JT - but its a morbid story here. Ex state monopoly, etc. etc. |
| 07:26.09 | JT | aiksa[LV]: they must be crazy |
| 07:27.10 | aiksa[LV] | JT, have heard about your internet prices. A friend of my went to study cinematography over there. Was pretty suprised. |
| 07:27.59 | jelly-bean | but is that 1.54mb per channel (e.g. with 24 channels potentially for data) or total? |
| 07:28.10 | aiksa[LV] | We can get connections for private purposes of 10Mbits/10Mbits, (non guaranteed though) for around 30 EUR a month without any bandwidth cap. |
| 07:28.17 | jelly-bean | i'm just surprised because t1 is still so pricey |
| 07:28.35 | jelly-bean | but the 8mb cable is just $62.50/mo |
| 07:28.42 | JT | jelly-bean: 64000bit/s per channel |
| 07:29.06 | JT | cable infrastructure is contented and unreliable |
| 07:29.10 | JT | of course it's cheap |
| 07:29.54 | jelly-bean | well they're offering it as a business-class solution |
| 07:29.57 | JT | aiksa[LV]: i can get Internet bandwidth for about $1.5/GB in the datacentre, that's cheap for datacentre bandwidth with no minimum commitment |
| 07:30.02 | JT | jelly-bean: well it's not |
| 07:30.28 | Strom_C | jelly-bean: 1.544 megabits -- 24 channels at 64kbps each |
| 07:30.40 | *** join/#asterisk LoneShadow (n=a@c-76-103-55-28.hsd1.ca.comcast.net) |
| 07:30.50 | Strom_C | jelly-bean: but seriously, don't go with the cable company for this stuff |
| 07:31.02 | Strom_C | they don't know what they're doing |
| 07:31.17 | JT | Strom_C: +8kbit/s sync |
| 07:31.28 | Strom_C | well it's framing, not sync |
| 07:31.31 | Strom_C | but anyway |
| 07:31.42 | JT | you know what i mean |
| 07:31.58 | JT | E1s are much simpler, 32 * 64 = 2.048Mbit/s ;) |
| 07:32.00 | *** join/#asterisk redback (n=noname@mail.datadream.co.uk) |
| 07:32.14 | Strom_C | JT: yeah, but counting in multiples of 31 isn't fun :P |
| 07:32.24 | JT | why would you do that? |
| 07:32.48 | Strom_C | I called up the cable company on behalf of a client who needed a mapping of what DNIS was programmed to each DID, and the technician on the other end asked "What's Geenis?" |
| 07:32.59 | JT | it's multiples of 30, and counting in multiples of 30 is much easier than counting in multiples of 23 |
| 07:33.12 | Strom_C | JT: in asterisk, the signaling channel is also counted |
| 07:33.14 | Strom_C | so 30 + 1 |
| 07:33.34 | JT | sure |
| 07:33.43 | JT | if you look at it that way :) |
| 07:33.53 | Strom_C | but I'm sure if I worked with E1 all the time it would make perfect sense to me too |
| 07:34.18 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 07:36.15 | aiksa[LV] | JT, still pricing by Gbit seems strange by our local pricing policies. |
| 07:36.36 | JT | aiksa[LV]: it's available per megabit or per GB |
| 07:36.44 | JT | and a few other schemes |
| 07:37.00 | *** join/#asterisk af_ (n=getsmart@88-149-230-31.dynamic.ngi.it) |
| 07:37.18 | Strom_C | a summary of the last half hour, for those who just joined: |
| 07:37.32 | Strom_C | "Boy, you sure have weird telecom products and services in [other country]" |
| 07:37.41 | aiksa[LV] | :) |
| 07:38.09 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:39.34 | *** join/#asterisk TeraFlood (i=grikis@in.delfi.lv) |
| 07:39.52 | cjk | hi; does asterisk depend on reverse dns lookups? if so, how can i disable thel |
| 07:42.18 | *** join/#asterisk steliosk (n=Stelios@athedsl-284816.home.otenet.gr) |
| 07:44.46 | LoneShadow | are there good free UK sip providers like ipkall ? |
| 07:45.09 | Strom_C | "good" and "free" almost never go together in telecom |
| 07:45.56 | LoneShadow | well something simillar to ipkall :) |
| 07:46.14 | JT | free sounds like a bad business proposition in telecoms |
| 07:47.40 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 07:52.03 | *** part/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 07:52.36 | *** join/#asterisk gr0mit (n=tim@CSC.ge4-0-0.401.ar1.BBS1.gblx.net) |
| 07:55.06 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 07:55.27 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 08:05.06 | *** join/#asterisk E-bola (n=bola@cpe-76-179-4-233.maine.res.rr.com) |
| 08:09.52 | L|NUX | can some one help me with DUNDi |
| 08:09.53 | L|NUX | please |
| 08:12.05 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 08:18.11 | tzafrir | L|NUX, try to be more specific |
| 08:18.15 | dandre | hello |
| 08:18.15 | jbot | hey |
| 08:18.24 | L|NUX | tzafrir: hold let me do pb |
| 08:18.34 | tzafrir | (and no, I really don't know much about this Dundi guy ;-) |
| 08:19.19 | dandre | astmanproxy doesn't understand passwords with mixed case letters. Is this a known issue? |
| 08:20.32 | *** join/#asterisk paulproteus (n=paulprot@wide-rose.makesad.us) |
| 08:20.49 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
| 08:21.02 | L|NUX | tzafrir: http://rafb.net/p/k6rMZV72.html |
| 08:21.21 | L|NUX | tzafrir: but might be you can help :) |
| 08:22.55 | L|NUX | i am registered on box which have AAA.AAA.AAA.AAA and calling on box which have ip BBB.BBB.BBB.BBB |
| 08:22.59 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:23.34 | tzafrir | just to focus your question: do you think that the dundi query was successful and the issue is with the SIP authentication? |
| 08:23.41 | L|NUX | yes |
| 08:23.49 | trnzmeta | guys, can you connect two asterisk boxes to voi pprovier with same account details? |
| 08:23.54 | L|NUX | might be sip is an issue |
| 08:24.03 | tzafrir | In that case leave dundy out of it |
| 08:24.18 | tzafrir | you can try a direct SIP call |
| 08:24.25 | L|NUX | okay hold |
| 08:25.35 | tzafrir | trnzmeta, for outgoing calls only ? also for incoming calls? |
| 08:25.59 | trnzmeta | just outgoing |
| 08:26.59 | *** join/#asterisk jarod14 (n=jarod14@LMontsouris-152-63-1-19.w80-12.abo.wanadoo.fr) |
| 08:34.28 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 08:36.12 | *** part/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 08:43.58 | *** join/#asterisk Asterisk_new (n=ijm@82-171-224-8.ip.telfort.nl) |
| 08:44.01 | Asterisk_new | hello |
| 08:44.01 | jbot | what's up, asterisk_new |
| 08:44.08 | Asterisk_new | May i ask a question please? |
| 08:44.40 | Asterisk_new | <PROTECTED> |
| 08:45.33 | Asterisk_new | the command i use is: Dial,SIP/MYNUMBER/0031612345678 |
| 08:45.37 | Asterisk_new | and it's working! |
| 08:46.03 | Asterisk_new | but it's quiet the first 4 seconds |
| 08:46.17 | Asterisk_new | can i add an extra ringing when dialing my mobile phone?? |
| 08:50.47 | MatBoy | is away: MatBoy Hides ;) |
| 08:51.33 | dandre | tzafrir: astmanproxy doesn't understand passwords with mixed case letters. Is this a known issue? |
| 08:52.04 | tzafrir | dandre, I don't know astmanproxy well |
| 08:52.10 | tzafrir | But what do you mean? |
| 08:52.17 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:52.24 | tzafrir | Aren't passwords case sensitive? |
| 08:52.38 | dandre | sorry I though you were the maintener |
| 08:52.39 | MatBoy | is back (gone 00:01:52) |
| 08:53.50 | dandre | yes in the manager they are but if I specify a mixed case password in astmanproxy.conf, it is passed lowercased to the manager |
| 08:54.01 | *** join/#asterisk steliosk (n=Stelios@athedsl-284816.home.otenet.gr) |
| 08:54.01 | *** join/#asterisk Zorix (n=Brandon@c-98-208-144-134.hsd1.fl.comcast.net) [NETSPLIT VICTIM] |
| 08:54.01 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) [NETSPLIT VICTIM] |
| 08:56.51 | Asterisk_new | does someone knows how to add an extra ringing on transfer ?? please |
| 08:57.54 | Asterisk_new | i'm searing for something like this: |
| 08:57.54 | Asterisk_new | Ring Back is the simulated ring a caller hears when they are being transfered to a extension. |
| 09:02.08 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 09:04.55 | *** join/#asterisk kannan (n=admin@121.243.115.129) |
| 09:05.34 | kannan | hellol, I am tring to register a grandstream 2020 model with my asterisk, but am not able to get it. other phones register fine with the same user details |
| 09:05.40 | Strom_C | Asterisk_new: what do you mean "extra ringing on transfer"? |
| 09:05.50 | Strom_C | ~gs |
| 09:05.50 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 09:06.19 | kannan | heh |
| 09:06.41 | kannan | how about polycom , i was given this GS as a sample by a vendor |
| 09:06.49 | Strom_C | I love polycom |
| 09:07.02 | Strom_C | the configuration can be a touch hairy, but the phones are solid |
| 09:07.02 | kannan | kewl then |
| 09:07.03 | *** join/#asterisk km2 (n=x@c-24-23-252-175.hsd1.ca.comcast.net) |
| 09:07.12 | kannan | hairy , on o |
| 09:07.15 | kannan | oh no |
| 09:07.23 | tzafrir | dandre, is there any place with some bug reports and fixes for astproxyman? |
| 09:07.34 | kannan | there is no help no the GS websites also |
| 09:08.00 | kannan | on voip wiki also, . i saw screen shots in asteriskguru and followed these, but the phone is not registering |
| 09:08.04 | tzafrir | maybe it would be worth opening a sourceforge project to collect the patches and bug reports for starters |
| 09:08.12 | Asterisk_new | Strom_C: my house phone is ringing 20 seconds |
| 09:08.52 | Asterisk_new | after that the caller is being "transfered" to my mobile phone |
| 09:09.20 | kannan | brb |
| 09:09.38 | Asterisk_new | but it's about 4-5 seconds quiet (the caller doesn't hear anything) |
| 09:10.03 | Asterisk_new | and after that my cell phone is ringing (and the caller hears the ringtone again) |
| 09:10.24 | Asterisk_new | can i add an extra ringing to break the silence ? |
| 09:12.54 | *** join/#asterisk steliosk (n=Stelios@athedsl-284816.home.otenet.gr) |
| 09:12.54 | *** join/#asterisk Zorix (n=Brandon@c-98-208-144-134.hsd1.fl.comcast.net) [NETSPLIT VICTIM] |
| 09:12.54 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) [NETSPLIT VICTIM] |
| 09:13.47 | Strom_C | Asterisk_new: probably not, unless you want to kill your call progress information from the network |
| 09:15.02 | *** join/#asterisk TrentCreek (n=trent@cpe-70-116-111-122.rgv.res.rr.com) |
| 09:15.05 | dandre | I don't know, from debian/changelog: |
| 09:15.06 | dandre | <PROTECTED> |
| 09:17.19 | *** join/#asterisk Penol (n=Penol@195.204.248.62) |
| 09:17.46 | tzafrir | dandre, the 1.2 package? |
| 09:18.19 | Penol | Will this extenstion work http://pastebin.no/6016 ? |
| 09:18.38 | kannan | Strom_C, any idea what the proxy-require filed is in the GS settings? |
| 09:19.04 | Strom_C | not without some context, no |
| 09:20.05 | Penol | Will this extenstion work http://pastebin.no/6016 ? |
| 09:20.32 | Asterisk_new | Strom_C: thank you for your help |
| 09:21.02 | TeraFlood | question? I have situation when asterisk server will make brige sip calls width Siemens Hipass8000 so how much one server - 2gb ram, duo core will can handle calls |
| 09:21.05 | dandre | from what I have downloaded (1.21 sourcetree) |
| 09:21.35 | dandre | do you know a better supported manager proxy tzafrir? |
| 09:24.22 | *** join/#asterisk Dr-Linux (n=somethin@117.20.21.66) |
| 09:24.26 | Dr-Linux | Hi all |
| 09:24.59 | Dr-Linux | i want encrypted password in voicemail.conf , i googled it but no help, anyone any idea for this? |
| 09:26.24 | TeraFlood | Dr-Linux U can try write some script and work width AGI |
| 09:29.09 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 09:29.19 | yang | tzafrir: could you tell me, how could I insert/delete a string in the dialplan ? Like I dial 00386123456 and the output should change into 0123456 , is it possible ? |
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| 09:30.12 | tzafrir | yang, temporarily? "dialplan add" (or "add" in 1.2) |
| 09:30.30 | tzafrir | You can also use the command "originate" |
| 09:30.36 | Dr-Linux | TeraFlood: does Asterisk provides some way to do that? |
| 09:31.09 | Dr-Linux | TeraFlood: if NO then is there any script already available to encrypt the passwords in voicemail.conf ? |
| 09:31.49 | yang | tzafrir: but I know how to cut the lines like 00386. then here comes {EXTEN:5}, but how to apend a 0 after? |
| 09:31.59 | Dr-Linux | tzafrir: i'd also like to have your comments on my question. Thanks |
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| 09:35.06 | tzafrir | ${EXTEN:5}0 ? |
| 09:35.12 | tzafrir | yang ==^ |
| 09:36.10 | tzafrir | Dr-Linux, not really sure. Too many things read voicemail.conf directly |
| 09:36.48 | trnzmeta | yeah baby, migration finished |
| 09:37.10 | tzafrir | Also: you mean "hashed password". Encrypted password does not make sense normally |
| 09:39.35 | TeraFlood | Dr-Linux the passwords in asterisk *.conf files always in plain txt. so when U conect to voicemail U can make AGI or other script witch will work for autorization from extension.conf - so also U can meke script and users&password take from SQL |
| 09:40.35 | aiksa[LV] | anyone here ever messed around with chan_alsa or chan_oss? The question is - when I dial chan_alsa will it automatically "Answer" the call? |
| 09:42.37 | aiksa[LV] | ok. my bad. It has autoanswer option in configs. case solved |
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| 09:54.27 | Dr-Linux | TeraFlood: was thinking to protect the voicmail.conf file not sure yet HOW |
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| 10:05.24 | TeraFlood | Dr-Linux 1st why - also U can try like http://www.voip-info.org/wiki/view/Asterisk+sip+md5secret |
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| 10:08.26 | klimonso | whenever i change the port of iax, if i restart it goes back to default port, how can i change it permenant ?? |
| 10:09.27 | klimonso | whenever i change the port of iax, if i restart it goes back to default port, how can i change it permenant ?? |
| 10:09.48 | klimonso | is there anyone alive here? |
| 10:10.08 | RoyK | waves |
| 10:10.35 | RoyK | klimonso: if you change it in iax.conf and restart and it "goes back to default", something is indeed wrong |
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| 10:15.24 | aiksa[LV] | klimonso: you are using bindport directive? |
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| 10:17.14 | klimonso | yes |
| 10:17.19 | klimonso | what else i can do so it stays |
| 10:17.29 | klimonso | coz port 4569 is blocked in my country |
| 10:18.42 | aiksa[LV] | klimonso: hmmm. but it stays the same in config file right? |
| 10:19.57 | yang | tzafrir: thanks |
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| 10:20.00 | klimonso | i tried editing it in ssh and i tried editing it in trixbox maint |
| 10:20.13 | klimonso | and when i reboot it goes back to its original state |
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| 10:21.01 | aiksa[LV] | klimonso: ok, so its not asterisk issue |
| 10:21.10 | aiksa[LV] | but rather configuration file override |
| 10:21.31 | aiksa[LV] | while I suppose this is more appropriate question for #trixbox forum. |
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| 10:24.11 | Faustov | hi guys, i have 2 register lines in my sip.conf: register => user1:pass1@sip.provider.com/extension1 and register => user2:pass2@sip.provider.com/extension2 (2 users, same provider) - whenever someone dials the number given by the first registry, it shows up in the console as Executing [extension@context:1] MeetMe("SIP/USER2-007f8510", "ext|Mix") in new stack |
| 10:24.21 | Faustov | is this a bug? it should definitely be USER1 |
| 10:28.38 | aiksa[LV] | klimonso: also try to look into your /etc/rc.d routines perhaps there is something which restores the old config |
| 10:31.21 | aiksa[LV] | Faustov: you could disable user1 and then look if call still comes in |
| 10:31.37 | aiksa[LV] | if yes then provider is sending it through user2 |
| 10:31.57 | aiksa[LV] | also take a look at what [user1] and [user2] records you have in your sip.conf |
| 10:32.54 | aiksa[LV] | because IMHO SIP/XXX doesnt correspond to user with username XXX, but rather to the [XXX] |
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| 10:36.21 | jeremy_g | ev |
| 10:40.37 | Faustov | aiksa[LV]: it does come in anyways, because the users get connected to that extension, but the logs say user2 is handling the call, while it's user1 |
| 10:41.27 | Faustov | i'll try disabling it later |
| 10:41.58 | aiksa[LV] | take a look at users definition portion in sip.conf with names [user1] and [user2] |
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| 10:42.12 | aiksa[LV] | this is where asterisk will decide to which user the call belongs to |
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| 10:47.35 | Faustov | aiksa[LV]: for local calls this is fine, however with registered public numbers it doesn't |
| 10:47.43 | Azam | Hello my asterisk drop a call after around 30 seconds please help. I am using xlite behind nat |
| 10:48.51 | Faustov | aiksa[LV]: if it registers as account1, i'd expect it to use the context as account1 |
| 10:49.41 | aiksa[LV] | Faustov: I am not the author of chan_sip, nor have I examined it in very fine details, but how I understand the workings of this is as follows: |
| 10:50.02 | aiksa[LV] | register=> only tells the other party that we are available at following IP address |
| 10:50.23 | Faustov | hmmm |
| 10:50.48 | Faustov | question: should i create contexts named as the usernames for the register lines? |
| 10:51.03 | Azam | my asterisk drop a call after around 30 seconds please help. I am using xlite behind nat |
| 10:51.10 | oej | Faustov: You need to understand matching for incoming calls here |
| 10:51.26 | oej | Faustov: We match incoming calls either on the From: username part of the URI or the IP/Port of the sender |
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| 10:51.37 | oej | The register=> has nothing to do with the matching |
| 10:52.00 | aiksa[LV] | oej: thats what I was trying to explain |
| 10:52.21 | oej | ...and I know a bit about the source... :-) |
| 10:52.41 | aiksa[LV] | not ... "and" ... but ..."but" |
| 10:53.15 | Faustov | oej: ok, i see, but where can I do some matching then? |
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| 10:53.43 | aiksa[LV] | Faustov: in your sip conf. where you would have [user1] and [user2] sections |
| 10:53.47 | oej | If you register two accounts with the same provider, you end up doing that in the dialplan |
| 10:53.54 | oej | If it's a PSTN provider |
| 10:54.27 | oej | The [user] matching won't work, because the From: is the caller ID of the person calling the provider |
| 10:54.37 | Faustov | aiksa[LV]: ok, in those sections i have 2 contexts, one with ext1, other with ext2, so they don't even overlap |
| 10:55.07 | Faustov | but it still fails |
| 10:55.19 | aiksa[LV] | oej: wouldnt provider identify himself in sip when pushing the call to his pbx? |
| 10:55.38 | oej | Only with domain |
| 10:55.47 | oej | It depends on the provider of course |
| 10:56.04 | oej | Normally you will get a From: <callerid>@providersdomain |
| 10:56.06 | Faustov | oej: isn't the dialplan only for outgoing calls? |
| 10:56.19 | oej | Faustov: The dialplan is for all calls |
| 10:56.31 | oej | What's an outgoing call? |
| 10:56.34 | aiksa[LV] | Faustov: there aint difference between incoming and outgoing |
| 10:56.44 | aiksa[LV] | whats incomming on line is outgoing on other |
| 10:56.47 | aiksa[LV] | and vice versa |
| 10:56.48 | oej | The dialplan executes an INCOMING call (from outside or inside your company) and sets up OUTBOUND calls |
| 10:58.10 | aiksa[LV] | oej, gotta run. I`ll leave the explaining on this one to you. *grin* |
| 10:58.30 | Azam | <PROTECTED> |
| 11:03.01 | Faustov | oej: ok, here's what i came up with: exten => user1,1,Dial(SIP/ext1,,) |
| 11:03.04 | oej | Hangup on the 29th second... |
| 11:03.07 | Faustov | oej: does this look correct to you? |
| 11:03.24 | oej | Faustov: Didn't you give "extension1" in the register= statement |
| 11:04.09 | Faustov | oej: i do, but that one isn't going through the right set of extensions |
| 11:04.20 | Faustov | oej: now i have it like this: |
| 11:05.59 | Faustov | oej: registry => x:y@provider/extension; then [account1] context=stations; then in [stations] exten => myext,1,Dial(SIP/ext1,,) |
| 11:06.32 | oej | "myext" and "extension" is not the same. Why? |
| 11:06.39 | Faustov | but if i understand you correctly, there's no link between the registry /extension info and the [account]'s context |
| 11:06.57 | Faustov | i'm sorry, it's the same |
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| 11:08.10 | oej | The link is the IP address of the server |
| 11:08.25 | oej | YOu need to create a [peer] with host=<hostname or ip> of your provider, then set the context there |
| 11:08.38 | Faustov | oh |
| 11:08.51 | Azam | 0. |
| 11:08.55 | Faustov | will that work since asterisk is behind NAT? |
| 11:09.19 | oej | The sender's IP address will be the same, regardless of NAT |
| 11:10.52 | Faustov | should that [peer] you mention be actually the [account] of the sip provider? |
| 11:11.26 | Faustov | or something extra? |
| 11:13.11 | Faustov | because where would i have a reference to [peer]? |
| 11:17.36 | oej | Faustov: I guess you need to read some documentation. There's plenty of examples out there. |
| 11:18.14 | Azam | Can anyone help me please? my asterisk drops a call after around 30 seconds. I am using xlite behind na |
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| 11:19.02 | Penol | Will this extenstion work http://pastebin.no/6016 ? |
| 11:19.03 | Nobbie | heya =) |
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| 11:20.59 | deltaray2 | Just curious, have any of you made a IP Phone to IP phone over a very long distance, like from the US to China? How was the quality/delay? |
| 11:21.42 | Nobbie | deltaray2: there are so many determining factors, the only way to find out if it will work for you is to test |
| 11:22.03 | redback | deltaray2: I have done UK to NZ a few times with no probs but UK to UK with probs, many factors. |
| 11:23.20 | Nobbie | we do australia to South africa, minor intermittent problems |
| 11:23.43 | deltaray2 | Do you know if there was less delay than if it was a normal POTS connection? |
| 11:24.16 | deltaray2 | I am going to test, but I have to wait for the person to set it up, I was just curious to know before what other experiences are. Thanks. |
| 11:24.29 | Nobbie | hard to say, but normally on a POTS platform, the jitter is constant wheras with VoIP it will be varialbe |
| 11:30.46 | Nobbie | astdb seems to be a bottleneck on my PBX with ~400 extensions, ~60 concurrent calls. has anyone tried the sqlite replacement for astdb or some other alternative/optimization ? |
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| 11:52.32 | xenonex | hello, how add extension on any 6 digits? |
| 11:54.27 | kaldemar | xenonex: _XXXXXX |
| 11:55.29 | xenonex | thanks |
| 11:55.29 | jbot | xenonex: sure thing |
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| 12:00.37 | jblack | When running monitor, in and out isn't getting mixed together. Any suggestions? |
| 12:02.20 | jblack | Hmm, looks like I need the m option |
| 12:02.49 | jblack | Monitor(,,m) |
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| 12:40.32 | Azam | Can anyone help me please? my asterisk drops a call after around 30 seconds. I am using xlite behind na |
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| 12:42.45 | klimonso | is there any expert with freepbx? i need support pvt msg me please |
| 12:50.57 | tzafrir | Azam, please pastebin a trace from the Asterisk CLI |
| 12:52.01 | Azam | tzafrir, thanks for replying. just give me a minute |
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| 12:58.26 | mknerd | what are the advantages of having a multi-line SIP phone, can I not just park an unlimited amount of calls and get dialtone again? |
| 12:58.41 | Azam | tzafrir, http://pastebin.com/d7f0185b8 |
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| 12:59.13 | Azam | tzafrir, i must tell you that my asterisk is on a global ip and my xlite client is behind NAT |
| 13:00.23 | cjk | does anyone know is asterisk is doing reverse dns lookups? |
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| 13:03.12 | Curus | mknerd, it can be handy to have several accounts configured, and it's easier to let the phone handle 3-way conferences |
| 13:05.53 | a-s | When an agent of a queue is called, it cannot aswer when his telephone not ringing. Coulkd I fix this problem ? |
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| 13:15.50 | a-s | a telephone can answer a call just if it is ringing... If I answer and it wasn't ringing, then I get the message "busy".... |
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| 13:17.04 | deltaray2 | This is a long shot, but is there any way when forwarding a call that comes in from an outside line and then goes to an outside line, to have the call then offloaded to the phone system and not tie up your channels? Does that make sense? |
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| 13:19.25 | Fusoya | I'm hoping someone can help me: I need to find a way to cause Asterisk 1.2 to send "call end" event notifications to another system... If I were using 1.4, I think I could do it with AGI... but 1.2's docs read like it kills the script immediately on hang-up |
| 13:20.33 | *** join/#asterisk s0ck (n=m@unaffiliated/s0ck) |
| 13:20.52 | Fusoya | Is there some other way that I can call a script with channel/extension/callID on hangup? |
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| 13:21.37 | Rico29 | hi |
| 13:21.48 | Rico29 | I have a problem with thomson st2030S provisioning |
| 13:22.02 | Rico29 | what name must the defaut config file have ? |
| 13:22.19 | Rico29 | st2030s.inf ? |
| 13:23.50 | Katty | good morning! |
| 13:23.54 | Rico29 | hi |
| 13:23.54 | jbot | bonjour, rico29 |
| 13:24.10 | Rico29 | jbot> huh ? |
| 13:24.11 | jbot | Welcome to ICQ. |
| 13:24.26 | keith4 | what's good about it? |
| 13:25.15 | Rico29 | ? |
| 13:26.28 | keith4 | (the morning) |
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| 13:33.35 | a-s | there is no method to answer a call when the telephone is not ringing ? |
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| 13:34.36 | Nugget | how would that work? |
| 13:35.05 | mknerd | what are the advantages of having a multi-line SIP phone, can I not just park an unlimited amount of calls and get dialtone again? |
| 13:35.32 | Nugget | mknerd: among other things it can be useful to categorize/distinguish incoming calls |
| 13:36.28 | mknerd | my dialplan adds to the callerID to distinguish incoming calls from the call attendant, can you give me some other examples |
| 13:36.55 | Nugget | that must be a real pain in the ass for calling people back from the phone. :) |
| 13:37.02 | mknerd | ? |
| 13:37.21 | Nugget | you can't just call back based on the callerid |
| 13:37.41 | mknerd | you can't? why is that? |
| 13:37.47 | Nugget | you've changed it, right? |
| 13:38.00 | mknerd | just the name portion, and only added to it |
| 13:38.27 | Nugget | ah, so you're just (potentially) dropping some of the name portion. I guess that's a decent compromise if you have only one line. |
| 13:38.45 | mknerd | 3 lines, but they roll into each other |
| 13:39.04 | Nugget | I meant on the phone, silly. wasn't that the whole point of your question? |
| 13:39.36 | mknerd | well currently I have some polycom 501, 3 line phones, but I am just curious on what the advantages to getting 6 line apperance phones would be |
| 13:39.52 | mknerd | appearance even |
| 13:41.01 | mknerd | would I want to assign the same sip account to each line? or different ones .. I guess I am just curious as to how others are using them |
| 13:41.23 | Nugget | different ones. asterisk doesn't support multiple connections from the same sip account |
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| 13:41.54 | Nugget | generally people use them to differentiate between incoming calls, or do to speed dials (and/or presence detection) |
| 13:42.56 | mknerd | that is not true, I have 3 lines all set to a single sip account and it works |
| 13:45.24 | mknerd | looks weird on the phone though .. |
| 13:45.25 | mknerd | lol |
| 13:45.31 | mknerd | 202, 202, 202 |
| 13:45.32 | mknerd | hehe |
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| 13:48.35 | thepacmanfan | my sip phone is showing as a peer, but it's not in the registry. should that phone be able to place an outbound call? |
| 13:49.05 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.139) |
| 13:49.11 | *** join/#asterisk jpeeler (n=jpeeler@216.207.245.1) |
| 13:49.28 | *** join/#asterisk ddunavant (n=David@75.145.240.14) |
| 13:53.29 | *** join/#asterisk kensuke_ (i=be0210a1@gateway/web/ajax/mibbit.com/x-f86066f525a838d5) |
| 13:53.42 | kensuke_ | Hi |
| 13:53.43 | kensuke_ | :D |
| 13:55.09 | kensuke_ | a question....... for asterisk... kernel 64bits or 32bits... what is you recomendation...? |
| 13:55.31 | Curus | There's no reason to use 32 bits anymore |
| 13:55.41 | kensuke_ | Ok :D |
| 13:55.51 | kensuke_ | thanks |
| 13:55.51 | jbot | kensuke_: sure thing |
| 13:55.56 | Curus | There never was, really |
| 13:56.29 | coppice | especially in the days of 16 bit CPUs :-) |
| 13:56.56 | aiksa[LV] | coppice: you always manage to ruin the day :) |
| 13:57.13 | aiksa[LV] | 16 bit CPUs to perform 8 bit tasks ? |
| 13:58.04 | coppice | come on. we are talking about Unix systems. Unix was written for 16 bit machines, not 8 bit |
| 13:59.31 | *** join/#asterisk ming_zym (n=ming_zym@123.103.29.194) |
| 14:01.03 | Curus | Actually Unix was originally for an 18-bit CPU |
| 14:02.10 | thepacmanfan | a 16-bit CPU made by a 2-bit engineering team |
| 14:02.24 | coppice | hey, you're right. I forgot about that. the PDP-7 was like a PDP-10 cut in two, wasn't it :-) |
| 14:02.54 | Curus | I don't know, I've never played with a PDP |
| 14:03.23 | adr3nalin3 | Hey guys I am having trouble getting festival to work in asterisk. I have modified the festival.scm and reloaded the system. I hear no audio and I am seeing no error messages in the asterisk console. Any suggestions? |
| 14:03.44 | coppice | well, I never played with a PDP-7, but I've loathed a few PDP-10s and PDP-11s :-) |
| 14:03.49 | Faustov | a good manual on creating IVRs - could anyone point me? |
| 14:04.20 | adr3nalin3 | Faustov: http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu, http://www.voip-info.org/wiki/view/Asterisk+cmd+Record |
| 14:04.25 | aiksa[LV] | coppice: i was just kidding |
| 14:04.50 | adr3nalin3 | Faustov: actually second link was supposed to be -> http://www.voip-info.org/wiki/view/Asterisk+tips+phrase+recording+menu |
| 14:04.55 | aiksa[LV] | giving a reference to what thepacmanfan broke down to two steps |
| 14:05.03 | Faustov | thanks adr3nalin3 |
| 14:05.03 | jbot | my pleasure, Faustov |
| 14:05.26 | tzanger | thepacmanfan: I thinkt he full quote is "Windows is a 32-bit extension to a 16-bit graphical shell for an 8-bit operating system originally coded for a 4-bit microprocessor by a 2-bit company that can't stand 1 bit of competition." |
| 14:05.33 | coppice | everyone gets so "power of 2" about word length these days. pretty much the only place that still breaks free is DSPs |
| 14:05.57 | *** join/#asterisk redback (n=noname@mail.datadream.co.uk) |
| 14:05.57 | tzanger | coppice: I thought anything to do with floating point was fully-free of powers of two as well |
| 14:06.03 | thepacmanfan | tranger, someone's gonna have to work 64-bit into that soon |
| 14:06.37 | coppice | tzanger: but they don't put it on a nice 36 bit word like a PDP-10 did, do they? |
| 14:07.10 | coppice | the first processor I developed was 24 bit. I'm a free spirit :-) |
| 14:07.12 | Curus | tzanger: Everyone does 64-bit doubles these days. Even Intel has given up on 80-bit except for backwards compatibility |
| 14:07.15 | adr3nalin3 | no problem Faustov |
| 14:07.40 | coppice | they only did 80 bit within the floating point engine |
| 14:08.01 | redback | I have set 'queue_members => mysql,asterisk,queue_members' in 'extconfig.conf' but when I use the 'AddQueueMember' command within ael it does not add the member into the table. Is this by design? |
| 14:09.25 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 14:10.25 | thepacmanfan | have any of you used asterisk-gui to configure sip phones? |
| 14:10.40 | adr3nalin3 | thepacmanfan: yes |
| 14:11.14 | adr3nalin3 | thepacmanfan: more like to configure the users for the phones all the actual phone config is done on the phone |
| 14:11.47 | thepacmanfan | adr3nalin3: a little sip config needs to be done on the asterisk side... can i do all that under Users in asterisk-gui |
| 14:11.52 | thepacmanfan | > |
| 14:11.54 | thepacmanfan | err.. ? |
| 14:11.57 | adr3nalin3 | yep |
| 14:12.22 | thepacmanfan | dangit... well, my 7960 is showing up as a sip peer, but not in sip registry. |
| 14:13.15 | *** join/#asterisk intralanman (n=lanman@209.85.58.2) |
| 14:13.23 | Curus | So, what is the scoop on DAHDI? |
| 14:13.37 | thepacmanfan | in *-gui i set the Name to the Auth Name on the phone, and i set the password to the Auth Password on the phone. |
| 14:13.54 | coppice | and when does MAHMI follow it? |
| 14:13.59 | Curus | Indeed |
| 14:15.06 | thepacmanfan | OTOH, it looks like the only reason that phone was showing up as a peer was due to my manual edit of sip.conf |
| 14:15.10 | rob0 | She never follows, she leads. |
| 14:15.35 | thepacmanfan | it's like asterisk-gui isn't even updating sip.conf, or even extensions.conf |
| 14:16.32 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 14:16.45 | Curus | Inquiring minds want to know |
| 14:18.49 | *** join/#asterisk supjigator (n=shanebur@152.53.16.10) |
| 14:19.54 | thepacmanfan | hmm... looks like asterisk-gui is putting everything in users.conf |
| 14:20.20 | thepacmanfan | is that good enough for SIP registration, or do i need the info in sip.conf too? |
| 14:25.48 | Curus | Perhaps sip.conf includes users.conf? |
| 14:26.44 | *** join/#asterisk mackes-Office (n=root@74.10.229.35) |
| 14:26.59 | *** join/#asterisk af_ (n=getsmart@88-149-230-31.dynamic.ngi.it) |
| 14:27.03 | mackes-Office | Good Morning Everyone |
| 14:28.02 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177585339.dsl.bell.ca) |
| 14:29.28 | thepacmanfan | ah, wait a minute. i bet it's my firewall. |
| 14:29.34 | *** part/#asterisk kensuke_ (i=be0210a1@gateway/web/ajax/mibbit.com/x-f86066f525a838d5) |
| 14:29.45 | thepacmanfan | what ports need to be open to allow a sip registration to happen? |
| 14:30.27 | rob0 | /topic |
| 14:30.49 | *** join/#asterisk rafaelrmrr (n=rafael@140.203.16.201.dekanet.com.br) |
| 14:31.16 | mackes-Office | That is a good question |
| 14:31.31 | mackes-Office | I have opened 5060, and 10000-20000 |
| 14:31.37 | rob0 | but anyway, if the client is registering with you, you need to open the SIP port (udp/5060) or the port the client is expecting to use. |
| 14:32.08 | thepacmanfan | everything is at defaults, so 5060... mackes, why the 10000-20000? |
| 14:32.49 | rob0 | whose registration is failing, yours or a SIP client? |
| 14:33.07 | thepacmanfan | a SIP client |
| 14:33.27 | mackes-Office | I am not sure the of the exact technical reason, but I am sure you need them for the call to have two way audio? |
| 14:33.33 | mackes-Office | rob0, do you agree? |
| 14:35.26 | tzafrir | thepacmanfan, registration is SIP, hence (in the case of Asterisk) UDP port 5060 |
| 14:35.32 | Curus | mackes-Office: On a lucky day the connection will start from the inside, and then you don't need to open the ports (assuming they are open outgoing) |
| 14:35.34 | *** join/#asterisk Skarmeth (n=Skarmeth@201009042244.user.veloxzone.com.br) |
| 14:35.56 | cjk | hi, asterisk doesnt seem to work when it has no internet |
| 14:35.58 | tzafrir | 10000-20000 might be used for the RTP payload managed by SIP sessions |
| 14:36.01 | Curus | But like all things NAT, things work sometimes. |
| 14:36.11 | mackes-Office | Hmmm Interesting. |
| 14:36.28 | *** join/#asterisk xenonex (n=xenonex@89.218.233.68) |
| 14:36.34 | mackes-Office | So when should ports 10000-20000 be forwarded. |
| 14:36.45 | thepacmanfan | tzafrir, so basically, my asterisk server needs 10000-20000 open |
| 14:37.00 | tzafrir | thepacmanfan, not for registration |
| 14:37.08 | Curus | mackes: When you can. |
| 14:37.15 | thepacmanfan | but for calls... ok. |
| 14:37.19 | Curus | mackes: If you can't, it'll possibly work anyway |
| 14:37.43 | Curus | Of course it's also a huge range of ports to open, so a firewall becomes somewhat silly. |
| 14:37.53 | Curus | Just hope nothing else uses a port in that range |
| 14:38.06 | mackes-Office | Ahh. So best practice yes, but if that NAT is held open via registration, then no. |
| 14:38.09 | Curus | (Or shrink it in rtp.conf) |
| 14:38.40 | thepacmanfan | i'm not dealing with NAT, just the default iptables config in centos with server options |
| 14:38.42 | Curus | Registration doesn't open the RTP ports. The SIP packets might, if the NAT knows about the SIP protocol |
| 14:39.46 | Curus | But then if the NAT knows about SIP, and the phone knows about NAT, they both try to fix up addresses and things break. Hello ZyXEL. |
| 14:40.30 | thepacmanfan | hmm... i opened 5060/udp, and my phone still won't register. |
| 14:40.45 | Curus | I'd say switch to IPv6, but everyone uses filters there too, so that doesn't help |
| 14:41.19 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 14:41.46 | Curus | thepacmanfan: Try tcpdump -nieth0 port 5060, see if any packets come from the phone. Or sip debug ip <phoneip> |
| 14:42.59 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 14:43.37 | Curus | tcpdump sees the packets before iptables does, sip debug in asterisk sees them afterwards |
| 14:44.12 | thepacmanfan | any reason i'd need to use -v with tcpdump |
| 14:44.15 | thepacmanfan | ? |
| 14:44.31 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:44.31 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:45.53 | Curus | Don't, unless you add -s0 too, or you'll see broken SIP packets because tcpdump doesn't fetch enough |
| 14:48.03 | thepacmanfan | hmm |
| 14:48.22 | thepacmanfan | tcpdump isn't seeing anything on 5060. |
| 14:48.41 | Curus | Then your problem isn't iptables (or your interface isn't eth0) |
| 14:49.18 | thepacmanfan | hmm... on my 7960, should http proxy addr be my asterisk server? |
| 14:49.33 | Curus | HTTP proxy? I doubt that |
| 14:50.03 | thepacmanfan | default router 1? |
| 14:50.27 | thepacmanfan | i mean, should i have to enter the host address of my asterisk server on my phone? i don't have that anywhere. |
| 14:51.07 | thepacmanfan | ip address seems to be the address of my phone. |
| 14:51.13 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 14:55.36 | Curus | Your asterisk server is registration server. Some broken phones require it as proxy, too |
| 14:56.22 | *** join/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 14:56.22 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 14:59.48 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:02.12 | *** join/#asterisk r0land (n=roland@193.227.191.91) |
| 15:02.15 | r0land | hello all |
| 15:02.24 | r0land | could someone help me with my sip to pstn setup plz! |
| 15:02.46 | r0land | im using spa3102 in between asterisk and the pstn line.. |
| 15:02.56 | r0land | though i cant call out nor recieve calls |
| 15:03.07 | r0land | even though postn interface is regsisted on asterisk |
| 15:05.13 | *** join/#asterisk Yourname` (n=chatzill@unaffiliated/yourname/x-837320) |
| 15:06.50 | fiddur | Hi folks. Is ip-telephone questions ok here? We're about to buy new phones for our office and have asterisk now (when previously only alcatel).. I'm looking at grandstream PBX-2000 for example, but on the grandstream-page on voip-info there are some comments about how bad it is etc, while others say it works well.. Are the bad comments just negative marketing for other phones or what? |
| 15:07.36 | rob0 | ~gs |
| 15:07.36 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 15:08.12 | Curus | We replaced GXP-2000 for our customers with Snom phones 2 years ago |
| 15:08.26 | sp00kz | ~snom |
| 15:08.26 | jbot | [snom] like all German products. High quality, but wacky engineering. :) |
| 15:08.30 | Yourname` | Hi. A call comes into 10, 5 seconds later, it is forwarded to ext 13. On 13, the call is picked up, but requires DTMF input. DTMF is being seen by Asterisk as is evident from CLI DTMF output, however, it doesn't seem to actually go to the caller. Why? http://pastebin.ca/1022811 |
| 15:08.34 | rob0 | ~phones |
| 15:08.35 | jbot | phones is probably http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 15:08.55 | *** join/#asterisk flush (n=SYN_SENT@ip216-239-83-43.vif.net) |
| 15:08.58 | fiddur | lol, ok... |
| 15:09.02 | r0land | hmm |
| 15:09.02 | Curus | Supposedly they have gotten better with time and firmware updates, but too late for us. |
| 15:09.16 | r0land | anyone could help out with sipura 3102 and asterisk ?! |
| 15:09.29 | rob0 | ~ask |
| 15:09.30 | jbot | methinks ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 15:09.40 | r0land | :) |
| 15:09.50 | thepacmanfan | Curus, will tcpdump on port 5060 actually capture udp traffic too? |
| 15:09.55 | fiddur | i've tried a polycom, and while it's good, it is far more expensive than most alternatives... If that's what it takes, we'll pay it, but it seems to be so many cheaper alternatives that are spoken well of (or just well marketed :P) |
| 15:09.59 | *** join/#asterisk dFence (n=chatzill@p54980C65.dip0.t-ipconnect.de) |
| 15:10.04 | Curus | thepacmanfan: Yes |
| 15:10.20 | *** part/#asterisk supjigator (n=shanebur@152.53.16.10) |
| 15:10.24 | Curus | You can do tcpdump udp port 5060 if you don't want traffic to tcp port 5060 |
| 15:10.34 | Curus | But you aren't likely to drown in TCP to 5060 |
| 15:10.35 | thepacmanfan | Curus: i'm getting NOTHING. :( |
| 15:10.35 | rob0 | Cheapest way to go is with ATA's and analog phones. |
| 15:10.51 | Curus | Anyway, I'll see if I can cut the handcuffs and stop being here against my will |
| 15:10.58 | r0land | i've added 2 extensions in sip.conf one for the softphone i have on my pc right now, and the other for the PSTN LINE interface on sipura 3102, i turned sip debugging on in the asterisk CLI and whenever i try to call out (i do so by calling the extension tht i assigned for the pstn interface) in the cli debugging gives me "03 service unavailable" |
| 15:10.59 | rob0 | but cheap might not be the best thing, in fact ... |
| 15:11.03 | rob0 | ~cheap |
| 15:11.04 | jbot | [cheap] 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. |
| 15:11.50 | Curus | You can't not be a cheapskate with Asterisk. A four-socket quadcore system plus phones is dirt cheap compared to proprietary solutions |
| 15:11.51 | fiddur | snom lies in the cheap interval as well, compared to polycom... but they're allright? |
| 15:12.09 | r0land | http://www.pastebin.ca/1022816 this is the error im getting in asterisk |
| 15:12.36 | fiddur | Curus: I know... we've gone over to asterisk because alcatel wanted another 50000 euro for a simple upgrade... |
| 15:12.54 | mackes-Office | I agree with Curus |
| 15:13.07 | mackes-Office | Great Phones, and a good server make Asterisk shine |
| 15:13.20 | rob0 | Curus, but that discounts the time investment. |
| 15:13.21 | mackes-Office | Low end phones make Asterisk look cheap |
| 15:13.29 | fiddur | but still, if a phone half the cost does the same job as e.g. polycom 601, why pay that? |
| 15:13.37 | Yourname` | Hi. I know my windows update settings are set to download but notify to install. When I boot up my PC, it shows the yellow icon meaning I;m supposed to install. But when I click it, it disappears and doesn't come back on. What do I do to instll the updates? |
| 15:13.53 | mackes-Office | Pickup the Polycom 320's for $85. |
| 15:13.59 | rob0 | looks up at the channel name |
| 15:14.02 | mackes-Office | They do the job, and they are very good phones |
| 15:14.14 | sp00kz | Polycom makes one of the best voip phones imho |
| 15:14.14 | r0land | rob0 ya i guess hes confused about it as well lol |
| 15:14.22 | r0land | rob0 so have any advice about my sip problem ? |
| 15:14.35 | redback | I have set 'queue_members => mysql,asterisk,queue_members' in 'extconfig.conf' but when I use the 'AddQueueMember' command within ael it does not add the member into the table. Is this by design? |
| 15:15.50 | thepacmanfan | i like the feel of the cisco 7960s we got for ~$125 apiece |
| 15:15.59 | r0land | so could some1 help with my sip prob! |
| 15:16.02 | thepacmanfan | of course, i'm having a heck of a time getting them properly set up for sip. |
| 15:16.14 | redback | thepacmanfan: heh - same one as your struggling with? |
| 15:16.20 | thepacmanfan | yep |
| 15:16.45 | thepacmanfan | they seem to be a well-made phone |
| 15:16.59 | thepacmanfan | fake dial tone sounds good, etc :) |
| 15:17.14 | Yourname` | Somebody? |
| 15:17.49 | fiddur | jbot mentioned Linksys SPA-9XX too; anyone here tried linksys SPA-921? |
| 15:18.08 | thepacmanfan | Yourname`: ##windows |
| 15:18.23 | redback | Yourname`: your asking the wrong question on the right channel, or the right question on the wrong channel |
| 15:18.32 | Yourname` | thepacmanfan: LOL, sorry.. scroll up. |
| 15:18.34 | Yourname` | Hi. A call comes into 10, 5 seconds later, it is forwarded to ext 13. On 13, the call is picked up, but requires DTMF input. DTMF is being seen by Asterisk as is evident from CLI DTMF output, however, it doesn't seem to actually go to the caller. Why? http://pastebin.ca/1022811 |
| 15:18.49 | Yourname` | Stupid complete. :) |
| 15:18.57 | rob0 | lol |
| 15:19.00 | thepacmanfan | hah |
| 15:19.12 | rob0 | I was fixin' t' berate Yourname` severely. |
| 15:19.30 | dFence | wow... now that i've tried 3 billing systems/plugins/bitches and nearly crippled my system in any possible way i think we just gonna charge 10bugs for each call ;D |
| 15:19.41 | rob0 | r0land: dunno, I guess your SIP client / sip.conf isn't set up right. |
| 15:19.42 | Yourname` | rob0: I did, and DTMF shows up in the CLI as detected. |
| 15:19.56 | Yourname` | rob0: Yet, it doesn't seem to be passed on to the caller. |
| 15:20.39 | r0land | rob0 if its not confi right.. how am i registered to it ? |
| 15:21.26 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 15:23.42 | *** join/#asterisk dmz (n=dmz@64.253.5.180.dyn-cm-pool75.pool.hargray.net) |
| 15:24.27 | dFence | hm.. my provider transmits a signal (probably via the d-channel/ISDN?) telling the PBX how to rate a call (local call 1unit/min, cell-phone 9units/min etc).. is there a way i can get a hold of that signal? (i am using chan_capi if that helps ;D) |
| 15:24.40 | *** join/#asterisk zbychuk (n=zbychuk@83-238-228-53.adsl.inetia.pl) |
| 15:24.41 | Bananaskin | thepacmanfan, u not tried sccp on the cisco's |
| 15:27.33 | *** join/#asterisk bmg505 (n=leon@196-209-78-202-tbnb-esr-2.dynamic.isadsl.co.za) |
| 15:28.59 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:29.20 | jblack | <PROTECTED> |
| 15:30.16 | jaytee | huh? |
| 15:32.21 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 15:32.21 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.4.20-rc3, 1.6.0-beta9 (2008/05/14) Asterisk 1.4.19.2 (2008/05/13), *-Addons 1.4.6 (2008/02/21) 1.6.0-beta3 (2008/04/02), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 15:34.27 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 15:36.30 | *** join/#asterisk dmz (n=dmz@64.253.5.180.dyn-cm-pool75.pool.hargray.net) |
| 15:37.16 | *** join/#asterisk ManxPower (n=manxpowe@37.sub-75-203-7.myvzw.com) |
| 15:40.26 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 15:40.26 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.4.20-rc3, 1.6.0-beta9 (2008/05/14) Asterisk 1.4.19.2 (2008/05/13), *-Addons 1.4.6 (2008/02/21) 1.6.0-beta3 (2008/04/02), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 15:42.04 | thepacmanfan | bananaskin: no, i haven't tried sccp. |
| 15:42.10 | adr3nalin3 | ManxPower: I am not sure it is a remote server |
| 15:42.26 | adr3nalin3 | I'm not seeing in error messages in the system log |
| 15:43.24 | adr3nalin3 | ManxPower: spoke to soon, I see errors now |
| 15:44.04 | *** join/#asterisk bootc (n=bootc@adsl1-p3903.ras.network-i.net) |
| 15:44.14 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 15:44.15 | bootc | hi folks |
| 15:44.27 | bootc | I'm trying to implement a hot-desking system which is working ok so far |
| 15:44.56 | bootc | but if you enable SIP overlap dialling on the phone it all falls apart, since it matches all extensions and then falls into the invalid context |
| 15:45.20 | ManxPower | adr3nalin3: you need to get it working outside of asterisk before you can expect to get it working inside of Asterisk |
| 15:45.45 | ManxPower | bootc: you should not need to enable overlap dialiing. Are you in some country with variable length phone numbers? |
| 15:46.13 | ManxPower | bootc: something is wrong with your dialplan. That should not be happening. |
| 15:46.18 | adr3nalin3 | ManxPower: yes I will troubleshoot my pulseaudio problem, I haven't ever had a problem with festival running on suse 10.3 kind of strange |
| 15:46.26 | bootc | there is that aspect, but there is also the case that I want people to be able to dial numbers without pressing OK on the phone |
| 15:46.50 | bootc | my code is (AEL): http://pastebin.com/m96de9b3 |
| 15:46.50 | ManxPower | bootc: a phone with a correct dialplan will not require them to press OK on the phone. |
| 15:46.53 | ManxPower | What phone are you using. |
| 15:46.57 | bootc | snom 300 |
| 15:46.59 | ManxPower | I can't help with AEL. |
| 15:47.05 | ManxPower | bootc: fix the phone dialplan then |
| 15:47.32 | bootc | what I get is http://pastebin.com/m43c0246a |
| 15:47.57 | keith4 | holy crap. i didn't think anyone actually used AEL |
| 15:48.18 | ManxPower | bootc: you are dialing extension 60? |
| 15:48.32 | redback | I thought ael was a replacement to .conf |
| 15:48.46 | ManxPower | redback: AEL is parsed into extensions.conf format on load. |
| 15:49.07 | ManxPower | All AEL is, is really a preprocessor to turn AEL into regular dialplan stuff. |
| 15:49.14 | keith4 | the default behavior of the snom 300 is to just sit there like an idiot, asking you to hit OK |
| 15:49.16 | *** join/#asterisk watchy (n=watchy@adsl-69-152-41-251.dsl.ltrkar.swbell.net) |
| 15:49.19 | *** join/#asterisk murdock_ut (n=chatzill@mail.kimballequipment.com) |
| 15:49.29 | watchy | ok i got the polycom intercom auto answer stuff working |
| 15:49.42 | watchy | but is there a way to call multiple phones and talk to them through it? |
| 15:49.46 | ManxPower | keith4: only if you are an idiot and did not configure the phone with the right dialplan |
| 15:49.46 | bootc | essentially the phone is in the 'hotdesk' context, and if you dial 60 it matches the extension and sends you off that way, then it goes to the hotdesk_loggedout context where '60' doesn't exist |
| 15:49.47 | keith4 | I think every other SIP phone that I have assumes you're done dialing and sends the number after a timeout |
| 15:49.57 | mackes-Office | Yep. |
| 15:50.03 | keith4 | ManxPower: notice where I said "default behavior" ? |
| 15:50.03 | mackes-Office | The Page command in Asterisk |
| 15:50.06 | ManxPower | bootc: it only does that if your dialplan is wrong. |
| 15:50.08 | bootc | I'd like it to keep sending "SIP/2.0 484 Address Incomplete" until it matches something in the destination context |
| 15:50.14 | watchy | any other way besides the page command? |
| 15:50.28 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 15:50.31 | *** part/#asterisk jelly-bean (i=user@63-76-119-176.directcom.com) |
| 15:50.31 | thepacmanfan | Curus: still around? |
| 15:50.37 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
| 15:50.44 | ManxPower | bootc: It DID match something in the destination context, it matched extension 60, priority 1 in the context hotdesk |
| 15:50.48 | bootc | I'm pressing the buttons to dial 6000, as soon as I press 60 it matches and starts dialling |
| 15:50.48 | mackes-Office | Well, you need multiple phones to answer- so they need to be put into some type of conference |
| 15:50.52 | mackes-Office | That is what page does |
| 15:50.53 | ManxPower | once it matches that your dialplan takes over |
| 15:50.56 | mackes-Office | Why no page? |
| 15:50.58 | watchy | yea thats what i was thinking |
| 15:51.03 | ManxPower | bootc: then don't have overlapping extensions! |
| 15:51.25 | bootc | well how else can I have my hot desking code if it doesn't match *everything* |
| 15:51.29 | keith4 | bootc: having extensions 60 and 6000 is a terrible idea |
| 15:51.34 | ManxPower | bootc: you think you have an overlap dial problem, you actually have a dialplan problem. |
| 15:51.38 | bootc | keith4: I don't, I only have 6000 |
| 15:51.46 | bootc | but my hot-desking code matches _X. |
| 15:52.05 | mackes-Office | Page is meetme- But with all of the phones muted |
| 15:52.14 | ManxPower | bootc: why? Do you really have to hotdesk every single possible number in the universe? |
| 15:52.22 | bootc | I'd like it to look-ahead to what context it will end up in and reply with a 'address incomplete' |
| 15:52.42 | watchy | ah when you use the page function it mutes the other phones |
| 15:52.49 | ManxPower | What context it ends up in is configured by context= in the sip.conf entry for that [device[ |
| 15:52.52 | bootc | I want the hot desking login to determine what context that phone ends up in |
| 15:53.01 | bootc | ManxPower: hotdesk |
| 15:53.34 | mackes-Office | yep |
| 15:53.35 | ManxPower | correct. The call will land in the [hotdesk] context. Once it gets there the phone has no more work to do and everything is handled in the dialplan |
| 15:53.39 | bootc | if I dial 6000 w/o SIP overlap dialling it works great |
| 15:54.02 | mackes-Office | Page is Meetme with muted phones- You can even turn the mute off with a switch |
| 15:54.07 | bootc | right, but I'd somehow like that context to pretend it doesn't exist unless it works out that number exists in the destination context |
| 15:54.11 | ManxPower | bootc: correct. That is because the phone sends 6000 enblock so the dialplan will never match 60 for that call. |
| 15:54.11 | mackes-Office | I do this with Polycom phones |
| 15:54.22 | bootc | ManxPower: correct |
| 15:54.41 | ManxPower | bootc: you cannot do what you want to do in the way you want to do it. |
| 15:55.06 | bootc | if I can't do it this way can I dynamically decide what context a phone lands in by default then? |
| 15:55.28 | bootc | I tried just changing the context in my SIP realtime DB but it's cached |
| 15:55.32 | ManxPower | You cannot dynamically decide what context the phone lands in. |
| 15:55.37 | ManxPower | It is simple as that. |
| 15:56.07 | ManxPower | Accept the call in the configured context, then send the call whereever you want to send it, but that is done in the dialplan. |
| 15:56.30 | rob0 | [russian-roulette] |
| 15:56.32 | ManxPower | This is the way almost everyone does it. |
| 15:56.46 | thepacmanfan | ok, i just noticed in tcpdump that my phone is sending traffic to asterisk if i try to place a call, but it never attempts to register itself.... |
| 15:57.00 | keith4 | what phone? |
| 15:57.05 | ManxPower | thepacmanfan: That's pretty common. Fix your phone config. |
| 15:57.16 | thepacmanfan | keith, cisco 7960 |
| 15:57.25 | *** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194) |
| 15:57.27 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.207.130) |
| 15:57.31 | keith4 | for some reason that I don't understand, many phones have a config option for "dial without registering" |
| 15:58.06 | keith4 | and, if I'm not mistaken, "answer without registering".. which seems fairly useless to me |
| 15:58.09 | ManxPower | the function of registration is to inform Asterisk the IP address of the sip user/pass. It does nothing else -- hence it has nothing to do with calls phone -> Asterisk, as asterisk does not need to know the IP address of the phone to ACCEPT calls from the phone, only needs to know that info if it wants to SEND a call to the phone. |
| 15:58.40 | ManxPower | keith4: many phones are on static IPs and so do not need to register because their IPs never change |
| 15:58.57 | keith4 | well, that's true |
| 15:58.57 | thepacmanfan | so i should be able to place outgoing calls even if the phone is not registered? |
| 15:59.10 | ManxPower | thepacmanfan: YES! |
| 15:59.17 | keith4 | but it could lead to a security nightmare |
| 15:59.24 | ManxPower | keith4: no it does not. |
| 15:59.25 | thepacmanfan | all my phones are going to be using static IPs, so i don't even need to use registration... |
| 15:59.25 | *** join/#asterisk mort_gib (n=mjensen@adsl-2-234.gibnet.gi) |
| 15:59.44 | ManxPower | thepacmanfan: then just host=theipofthephone for each [device] in sip.conf |
| 16:00.03 | thepacmanfan | sweet |
| 16:00.08 | ManxPower | you will still need the user/secret correct |
| 16:00.27 | bootc | ManxPower: either that or stop using overlap dialling |
| 16:00.37 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 16:00.38 | thepacmanfan | ok, so how do i open a port range in iptables? |
| 16:00.43 | ManxPower | bootc: if your dialplan was correct, overlap dialing would work. |
| 16:00.47 | ManxPower | thepacmanfan: see #linux |
| 16:01.06 | thepacmanfan | i need to open that whole port 10000 to 20000 for... uh... RTP? i'm guessing that's UDP? |
| 16:01.19 | thepacmanfan | *port range |
| 16:01.38 | rob0 | thepacmanfan, by default everything IS open. How to open it depends in part on how it was closed. I suggest disabling the firewall until you know what to do with it. |
| 16:01.39 | jblack | it's more manageable if you just open 2.5 * (max number of calls you expect) |
| 16:02.53 | thepacmanfan | rob0, my experience so far in this install of centos is that a number of things have to be opened, including 8088 and 5060 |
| 16:03.24 | thepacmanfan | jblack, the phones will sense the range that is open? |
| 16:03.50 | jblack | I can't speak as to the phones, just asterisk itself. |
| 16:03.55 | jblack | Its' easier if you turn redirect off |
| 16:04.24 | thepacmanfan | redirect? |
| 16:04.35 | rob0 | I'm talking about Linux defaults, not what firewall rules your distributor gave you. |
| 16:04.54 | thepacmanfan | ok. |
| 16:05.42 | [TK]D-Fender | "linux defaults" ... "distributor gave you"..... WTF? |
| 16:07.04 | [TK]D-Fender | RH distros offer a firewall installation option for which the correct answer is "none" after which you build your own. |
| 16:07.09 | bootc | can I use SIPPEER(foo,context)=blah_context to change the context? |
| 16:07.18 | ManxPower | bootc: NO! |
| 16:07.25 | ManxPower | You use a Goto like the rest of us |
| 16:07.44 | bootc | which breaks overlap dialling if I want to catch all calls from a phone |
| 16:07.49 | ManxPower | or include => of course. |
| 16:08.07 | [TK]D-Fender | bootc, Why would you be trying to change a SIP peers starting context from within the dialplan? |
| 16:08.13 | ManxPower | overlap dialing has nothing to do with your problem |
| 16:08.26 | ManxPower | your problem is overly broad pattern matches |
| 16:08.43 | bootc | [TK]D-Fender: if I, say, want to run 2 separate companies from one * with one set of phones |
| 16:08.51 | rob0 | "By default everything IS open." When iptable_filter initializes, nothing is blocked. |
| 16:08.52 | ManxPower | overlap dialing can EXPOSE the bugs in your design, but it won't cause it. |
| 16:08.54 | bootc | each user logs in and gets their own view of the phone system |
| 16:09.12 | bootc | 2000 to one user could be a completely different extension to 2000 for another user |
| 16:09.14 | bootc | that's the idea |
| 16:09.23 | ManxPower | bootc: The phones go into a single context, from that context you jump to the context for the correct company. |
| 16:09.37 | [TK]D-Fender | ^^^ |
| 16:09.40 | bootc | ManxPower: exactly what I'm trying to do |
| 16:09.46 | ManxPower | bootc: I already said you can't do what you want to do in the way you want to do it. |
| 16:09.47 | [TK]D-Fender | bootc, as ManxPower suggests... |
| 16:10.02 | bootc | but this is what I'm trying to do |
| 16:10.05 | [TK]D-Fender | bootc, single catch-all inbound, conditional goto |
| 16:10.19 | bootc | [TK]D-Fender: exactly what I _have_ |
| 16:10.25 | ManxPower | In theory you could set a global variable on login with the context you want the phone to go to, then use a Goto and look up the variable. |
| 16:10.41 | redback | I am setting up cdr_mysql and the uniqueid is not being set - according to http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql I need to set ASTCFLAGS+=-DMYSQL_LOGUNIQUEID at compile time. Do they mean compile time of asterisk or asterisk-addons |
| 16:10.53 | ManxPower | [TK]D-Fender: he's using _X. and that catches both 60 and 6000 and he does not want that. |
| 16:10.57 | Qwell | redback: addons |
| 16:11.20 | ManxPower | [TK]D-Fender: and he's using overlap dialing rather than the dialplan on the phone. |
| 16:11.20 | [TK]D-Fender | bootc, well... you COULD make a script that would change the sip.conf entry and issue a "sip reload". |
| 16:11.29 | ManxPower | [TK]D-Fender: He's pretty much doing everything he should not be doing. |
| 16:11.31 | redback | Qwell: just as I pressed enter I saw it there. And I did re-read it several times lol |
| 16:11.43 | [TK]D-Fender | bootc, perhaps realtime SIP peers would do that as well. |
| 16:12.34 | ManxPower | Goto(${${EXTEN}_CONTEXT},1,1) |
| 16:12.39 | bootc | [TK]D-Fender: great, but how? |
| 16:12.41 | [TK]D-Fender | ManxPower, there are a few ways around this as I suggested. A little kludgy, but largely effective |
| 16:12.57 | [TK]D-Fender | bootc, go make a scrip for your "login that will mod the config files. |
| 16:13.00 | Qwell | ManxPower: ${${EXTEN}_CONTEXT} ? |
| 16:13.09 | ManxPower | you do a Set(${${EXTEN}_CONTEXT=${LOGGEDIN_CONTEXT}],a) IIRC |
| 16:13.14 | Bananaskin | Hey guys have a weird problem on VM msgXXX.txt which appears to have wrong perms instead of 0755 has 0076, anyone seen this before ? |
| 16:13.19 | bootc | [TK]D-Fender: and then do a sip reload peers for it to take effect? |
| 16:13.33 | [TK]D-Fender | bootc, funny that sounds like what I just told you to do. |
| 16:13.46 | ManxPower | Qwell: you could use anything unique about the device. |
| 16:13.51 | redback | Qwell: any idea where to add it in the Makefile? |
| 16:14.26 | bootc | [TK]D-Fender: to be fair you didn't mention doing a sip reload :-P |
| 16:14.41 | ManxPower | bootc: how many times per hour will people be logging on / logging off? |
| 16:14.44 | [TK]D-Fender | <[TK]D-Fender> bootc, well... you COULD make a script that would change the sip.conf entry and issue a "sip reload" <------- |
| 16:14.47 | [TK]D-Fender | boot ^^ |
| 16:14.54 | [TK]D-Fender | bootc, O RLY? |
| 16:15.10 | bootc | ah didn't see that one above there :-P |
| 16:15.20 | bootc | this is indeed a big kludge though |
| 16:15.32 | ManxPower | pay attention, us old people don't have a lot of patience |
| 16:16.08 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-10-21-ndn-esr-2.dynamic.isadsl.co.za) |
| 16:16.23 | ManxPower | Qwell: My idea is to store the login/logoff information in a global variable unique to the device. EXTEN was prolly not the best choice. |
| 16:16.27 | *** join/#asterisk qdk_ (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 16:16.43 | coppice | get off ManxPower's lawn |
| 16:16.51 | [TK]D-Fender | bootc, well * can't do everything you imagine in an elegent way. Deal with it. |
| 16:17.15 | ManxPower | You could also store the state in AstDB, or any other database. |
| 16:17.28 | redback | ok - I really have no clue of where to add the line ASTCFLAGS+=-DMYSQL_LOGUNIQUEID in my Makefile - anyone able to enlighten me? |
| 16:17.50 | ManxPower | redback: we expect basic networking, linux, and dev experience before you use Asterisk. |
| 16:18.19 | ManxPower | If you don't have that experience, just look straight up. That's your "learning curve" |
| 16:18.39 | redback | ManxPower: I have most of those but not port development |
| 16:18.58 | [TK]D-Fender | ManxPower, frankly I don't see why soemone would have to add that anyways. Why isn't it logged by default? |
| 16:19.01 | b11d` | i believe there is linux support in #linux |
| 16:19.14 | Qwell | ##linux |
| 16:19.23 | redback | [TK]D-Fender: I completley agree |
| 16:19.40 | ManxPower | [TK]D-Fender: hotdesking is a very complex thing for something that sounds so simple. |
| 16:19.55 | redback | I'd rather ##FreeBSD - and will ask there - just thought you guys would know exactly what I mean |
| 16:20.05 | b11d` | redback... I use freebsd too :) |
| 16:20.05 | thepacmanfan | how much traffic should i expect between a phone and the server to place a call? |
| 16:20.05 | b11d` | welcome |
| 16:20.13 | redback | b11d`: :o) |
| 16:20.15 | [TK]D-Fender | ManxPower, well I just gave a pretty solid solution that even I could script..... |
| 16:20.18 | ManxPower | We thought about doing hotdesking, but our users all have memory loss (at least they appear to) and can never remember to log in and log off. |
| 16:20.34 | b11d` | watch out for fbsd 7 and asterisk.. they arent playing well right now |
| 16:20.40 | b11d` | stick with 6.2/6.3 |
| 16:20.45 | thepacmanfan | on port 5060 i'm getting one packet from the phone to the server, one back to the phone, and a third back to the server... that's it. |
| 16:20.58 | [TK]D-Fender | ManxPower, And then there is the way your client handles multiple line keys and... well we just won't got there, ok? ;) |
| 16:21.01 | thepacmanfan | i'm not getting a thing on ports 10000-20000 |
| 16:21.15 | redback | b11d`: I heard this, I wont be going 7 till 7.1 (except on desktop) - thanks for the heads up |
| 16:21.37 | b11d` | im about to downrev a test box to 6.2 to try to get the digium g729 codec working |
| 16:21.39 | ManxPower | [TK]D-Fender: for "hot desking" we just set up each line appearance for each different user. Not real hotdesking, but close enough for us. |
| 16:22.03 | bootc | so what are the drawbacks of issuing lots of 'sip reload peers' calls when people log in/out and I have ~100 phones? |
| 16:22.09 | ManxPower | b11d`: isn't it great that you are the only *BSD Asterisk user? 8-) |
| 16:22.23 | b11d` | i am SO not the only one |
| 16:22.25 | [TK]D-Fender | thepacmanfan, describe what it is you are attempting to have talk to *, and all of the networking in between. |
| 16:22.31 | b11d` | and im not ditching FreeBSD for any reason |
| 16:22.32 | ManxPower | bootc: you never answered my question so we don't know. |
| 16:22.41 | redback | ManxPower: come off it - loads of people use fBSD and * |
| 16:22.55 | ManxPower | redback: the poor misguided souls. 8-) |
| 16:22.56 | *** join/#asterisk bkw__ (n=brian@adsl-71-153-169-69.dsl.tul2ok.sbcglobal.net) |
| 16:22.58 | [TK]D-Fender | bootc, nothing that I can see.... should happen often enough for you to care about. |
| 16:23.01 | redback | wouldn't give up the stability for anything |
| 16:23.24 | bootc | well I suspect 100 people will log in when they get in in the morning between 8:30 and 9:05 (most and the end of that segment) |
| 16:23.34 | bootc | and then they'll all logout when they leave at 17:00 |
| 16:23.34 | b11d` | I just love FreeBSD.. i ditched linux back in 2001 and havent regretted it |
| 16:23.45 | bootc | with the odd login/out during the day |
| 16:23.45 | fiddur | I have an audio problem with my Polycom 301. When connected to one asterisk server, the audio is crappy, and connected to another it's great. I have compared the conf-files and I can't figure out what the difference is. They were installed differently; the asterisk it works with is branch 1.4 from about 1 week ago, the other was from asterisk-1.4.18 and later upgraded to 1.4-branch via svn. What should I look for? The bad calls are reported on CLI us |
| 16:23.57 | b11d` | redback.. there is an asterisk-bsd irc channel too.. #asterisk-bsd (its always SO dead though!) |
| 16:24.24 | Qwell | like BSd |
| 16:24.25 | Qwell | ducks |
| 16:24.30 | [TK]D-Fender | fiddur, whats the networking between each? What codec? maps BOTH out completely in a pastebin for us. |
| 16:24.31 | bootc | ManxPower: ^^^ |
| 16:24.31 | b11d` | hahah |
| 16:24.32 | [TK]D-Fender | ~pb |
| 16:24.33 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 16:24.34 | [TK]D-Fender | ^^^^^^^^^^^ |
| 16:24.34 | thepacmanfan | D-Fender: i've got a Cisco 7960 with the 8.2 SIP image, running through switches to Asterisk on Centos 5.1 with mostly-default iptables. the phone is not regestering at boot, but it's got a static IP, so it shouldn't matter. the phone is communicating to the server on port 5060 when i try to place a call, but i just get a fast busy signal, and i suspect the phone is generating it. |
| 16:24.57 | ManxPower | thepacmanfan: sip debug |
| 16:25.05 | redback | b11d`: dead because they are busy, not because we are the only ones :) |
| 16:25.18 | b11d` | agreed |
| 16:25.30 | [TK]D-Fender | thepacmanfan, probably because your dialplan or SIP peer isn't set right. You shouldn't be wasting timelike this. go look at the SIP DEBUG <------ |
| 16:25.35 | [TK]D-Fender | thepacmanfan, PASTEBIN is your friend. |
| 16:25.36 | fiddur | [TK]D-Fender: you meen the CLi-output on verbosity 3 or so? |
| 16:25.51 | ManxPower | I never could figure out this Stability Fanboy stuff. I've never had asterisk crash because of an OS problem. |
| 16:26.10 | [TK]D-Fender | fiddur, no, just the description of your config for BOTH setups o we can see the differences. |
| 16:26.19 | b11d` | yeah its really not a feature thing for me i guess.. i just prefer it.. |
| 16:26.22 | b11d` | in general, that is |
| 16:26.32 | ManxPower | and I've only ever had one actual bug with linux that impacted stability and that was with squid/reiserfs/directories with huge numbers of files. |
| 16:26.41 | thepacmanfan | manx, d-fender: i've never used sip-debug... does it echo debug data live to the CLI? |
| 16:26.49 | thepacmanfan | or do i need to hunt down a log file? |
| 16:27.01 | ManxPower | the yes, so make sure you have a big scrollback buffer |
| 16:27.09 | b11d` | with the amount of data a sip debug can generate, you WANT it going to a file :) |
| 16:27.16 | redback | ManxPower: I am not talking asterisk specificly but the various apps we use, and I just find it a nicer system to work on. |
| 16:27.27 | ManxPower | sip debug peer X is better, of course. |
| 16:27.28 | fiddur | [TK]D-Fender: which conf-files? codecs.conf is identical between the servers... the specific user in users.conf too |
| 16:27.33 | [TK]D-Fender | b11d`, for this no. |
| 16:27.38 | ManxPower | Asterisk has a codecs.conf? |
| 16:27.47 | ManxPower | fiddur: are you using a GUI Asterisk? |
| 16:28.08 | [TK]D-Fender | fiddur, just the GENERAL DESCRIPTION, not conf files yet. Where are these 2 servers? whats the networking between each and the phone. |
| 16:28.12 | Yourname` | Hi. A call comes into 10, 5 seconds later, it is forwarded to ext 13. On 13, the call is picked up, but requires DTMF input. DTMF is being seen by Asterisk as is evident from CLI DTMF output, however, it doesn't seem to actually go to the caller. Why? http://pastebin.ca/1022811 |
| 16:28.32 | fiddur | ManxPower: Yes, I started out with that, but have made a lot of changes by hand later. |
| 16:28.50 | ManxPower | fiddur: you would have to wipe your config files before we can really help you. |
| 16:28.55 | ManxPower | ~trixbox |
| 16:28.56 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 16:29.03 | ManxPower | ~freepbx |
| 16:29.04 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 16:29.15 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 16:29.45 | ManxPower | Yourname`: the rtT could be causing that |
| 16:29.46 | fiddur | [TK]D-Fender: The working server was placed on the same location as the one that is not working... so that was identical too |
| 16:29.56 | [TK]D-Fender | Yourname`, "requires DTMF" ? HUH? |
| 16:30.16 | [TK]D-Fender | fiddur, Ok, you are not listening, I really can't help you.... |
| 16:30.30 | thepacmanfan | cripes! it's my dialplan! |
| 16:30.41 | thepacmanfan | a call to ext 6000 is fine :) |
| 16:30.44 | Yourname` | ManxPower: You think the r, you mean? Isn't tT supposed to help? |
| 16:30.48 | thepacmanfan | and i hear cool music! :) |
| 16:31.00 | ManxPower | Yourname`: no, I meant what I said. |
| 16:31.06 | ManxPower | tT captures DTMF |
| 16:31.07 | *** join/#asterisk IOscanner (n=IOscanne@cpe-76-187-195-124.tx.res.rr.com) |
| 16:31.17 | ManxPower | you don't want it capturing DTMF, you want the DTMF to be sent as is. |
| 16:31.46 | ManxPower | I guess bootpc hot tired of us telling him he is wrong |
| 16:32.00 | Yourname` | ManxPower: Exactly. So if the call comes in and says please press 1 to be connected, I press 1.. and it connects be to somebody. Isn't that due to tT? |
| 16:32.21 | [TK]D-Fender | ManxPower, Well I hand fed him the answer on how to do it. Maybe he's figured he's done enough here... |
| 16:32.23 | ManxPower | Yourname`: not at all. |
| 16:32.29 | ManxPower | Yourname`: don't add options you don't understand |
| 16:32.41 | *** join/#asterisk skyphyr (n=quassel@92-234-8-140.cable.ubr07.hari.blueyonder.co.uk) |
| 16:32.50 | *** join/#asterisk iEatChildren (n=WaffleMu@asa.redglaze.com) |
| 16:32.52 | ManxPower | the press 1 to connect is done by your DIALPLAN |
| 16:32.52 | Yourname` | ManxPower: tT is for transferring, not DTMF. I understand.. but somehow I thought it's connected, but oh well. |
| 16:33.19 | ManxPower | you are not transfering anything anyway. |
| 16:33.28 | Yourname` | ManxPower: And I need the tT for the transfering too. So there's no way I can make the DTMF work without removing the tT? |
| 16:33.30 | ManxPower | not as far as Asterisk is concernes. |
| 16:33.40 | ManxPower | Yourname`: you know if you just tried it you would know by now. |
| 16:34.05 | *** join/#asterisk jsolis (n=jimmy@190.41.82.1) |
| 16:34.06 | Yourname` | ManxPower: lol i'm trying it.. but I'm wondering what if I need both the dtmf AND the transferring to work for different types of calls? |
| 16:34.11 | ManxPower | Yourname`: you only need Tt (DTMF transfer hack) if the phones you are using are too stupid to have their own transfer button |
| 16:34.18 | [TK]D-Fender | Yourname`, you make dtmf work by SETTING THE RIGHT MODE <- |
| 16:34.33 | ManxPower | Are the phones you are using too stupid to have their own transfer button? |
| 16:34.45 | [TK]D-Fender | ManxPower, transfer isn't the issue |
| 16:34.51 | jsolis | hey guy why dont work the zaptel.4.10.1 app_dial.c:1183 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown |
| 16:34.52 | *** join/#asterisk uluatu (n=deg@200.195.161.164) |
| 16:35.00 | [TK]D-Fender | ManxPower, his call comes in ASKING the receiver to acknowledge the call. |
| 16:35.04 | ManxPower | jsolis: you installed zaptel after Asterisk |
| 16:35.15 | Yourname` | [TK]D-Fender is right |
| 16:35.18 | jsolis | not before |
| 16:35.27 | [TK]D-Fender | Yourname`, so fix your bloody modes. |
| 16:35.35 | jsolis | first zaptel -> libpri -> asterisk |
| 16:35.42 | ManxPower | If asterisk does not see zaptel when you build it you won't get zaptel support. |
| 16:35.52 | ManxPower | jsolis: what happens when you load chan_zap.so in the CLI? |
| 16:35.59 | Yourname` | [TK]D-Fender: I wish I knew what the bloody modes where. |
| 16:36.05 | [TK]D-Fender | jsolis, PASTEBIN the complete CLI output of your failed call at verbose 10, and your zapata.conf and zaptel.conf |
| 16:36.13 | [TK]D-Fender | ~pb |
| 16:36.13 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 16:36.17 | jsolis | -- Reloading module 'chan_zap.so' (Zapata Telephony) |
| 16:36.21 | jsolis | <PROTECTED> |
| 16:36.24 | ManxPower | jsolis: does nit work now? |
| 16:36.25 | jsolis | [May 19 11:35:35] WARNING[7484]: chan_zap.c:11231 process_zap: Ignoring imidiate |
| 16:36.29 | jsolis | [May 19 11:35:35] WARNING[7484]: chan_zap.c:11231 process_zap: Ignoring signalling |
| 16:36.33 | redback | noooooooooooooooooooooooooooooooooo |
| 16:36.33 | [TK]D-Fender | ManxPower, he'd get a channel not implemented error if it didn't load... |
| 16:36.33 | jsolis | <PROTECTED> |
| 16:36.37 | jsolis | <PROTECTED> |
| 16:36.38 | ManxPower | jsolis: ignore the ignore message unless you are changing those options. |
| 16:36.41 | jsolis | <PROTECTED> |
| 16:36.44 | ManxPower | jsolis: do NOT flood the channel. |
| 16:36.45 | jsolis | [May 19 11:35:35] WARNING[7484]: chan_zap.c:11231 process_zap: Ignoring signalling |
| 16:36.49 | jsolis | [May 19 11:35:35] WARNING[7484]: chan_zap.c:11231 process_zap: Ignoring canreinvite |
| 16:37.01 | jsolis | [May 19 11:35:35] WARNING[7484]: chan_zap.c:11231 process_zap: Ignoring hasexten |
| 16:37.05 | jsolis | [May 19 11:35:35] WARNING[7484]: chan_zap.c:11231 process_zap: Ignoring hasiax |
| 16:37.09 | jsolis | [May 19 11:35:35] WARNING[7484]: chan_zap.c:11231 process_zap: Ignoring hassip |
| 16:37.12 | [TK]D-Fender | Qwell, ? |
| 16:37.13 | jsolis | [May 19 11:35:35] WARNING[7484]: chan_zap.c:11231 process_zap: Ignoring trunkname |
| 16:37.17 | jsolis | [May 19 11:35:35] WARNING[7484]: chan_zap.c:11231 process_zap: Ignoring trunkstyle |
| 16:37.21 | jsolis | <PROTECTED> |
| 16:37.25 | jsolis | <PROTECTED> |
| 16:37.29 | jsolis | not |
| 16:37.30 | ManxPower | jsolis: canreinvite, hasexten, hasiax, and hassip are NOT VALID zapata.conf options. |
| 16:37.33 | [TK]D-Fender | It'll be done in a sec anyways.... |
| 16:37.33 | jsolis | dont work |
| 16:37.33 | jsolis | when i use this sintaxis |
| 16:37.33 | jsolis | dial(zap/g1/${EXTEN}) |
| 16:37.33 | jsolis | but when i use dial(zap/1/${EXTEN}) |
| 16:37.33 | jsolis | work fine |
| 16:37.45 | Qwell | ~lart jsolis |
| 16:37.45 | jbot | pushes the wall down onto jsolis whilst whistling innocently |
| 16:37.48 | ManxPower | jsolis: looks like you don't have a group=1 defined |
| 16:37.56 | [TK]D-Fender | jsolis, You didn't set the GROUP for your zap channels. |
| 16:38.01 | ManxPower | jsolis: I'll bet this is a GUI install. |
| 16:38.25 | ManxPower | Qwell: can't you just kick/ban the flooders? |
| 16:38.37 | jsolis | this is my zapata.conf |
| 16:38.39 | Qwell | I could. but no |
| 16:38.42 | jsolis | [channels] |
| 16:38.44 | Qwell | unless he pasts now |
| 16:38.46 | jsolis | context=DID_trunk_1 |
| 16:38.47 | *** kick/#asterisk [jsolis!i=north@pdpc/sponsor/digium/Qwell] by Qwell (Qwell) |
| 16:39.15 | ManxPower | thank you qwell |
| 16:39.23 | ManxPower | next time maybe he'll use pastebin |
| 16:39.26 | *** join/#asterisk jsolis (n=jimmy@190.41.82.1) |
| 16:39.31 | Qwell | ~pb |
| 16:39.32 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 16:39.32 | ManxPower | jsolis: next time use pastebin |
| 16:39.38 | jsolis | soory |
| 16:40.11 | ManxPower | Once you get kicked by Qwell's Foot of Justice most people start doing what they are supposed to be doing. |
| 16:40.20 | [TK]D-Fender | jsolis, and we already told you what was missing. |
| 16:40.49 | tzafrir | <jsolis> [May 19 11:35:35] WARNING[7484]: chan_zap.c:11231 process_zap: Ignoring imidiate <=== typo |
| 16:41.08 | *** join/#asterisk EnoCix (n=jsloan@216.207.245.1) |
| 16:41.15 | ManxPower | I suspect he does not want immediate=yes anyway |
| 16:41.25 | *** join/#asterisk DJF5 (n=irc@84-105-201-37.cable.quicknet.nl) |
| 16:41.28 | Yourname` | ManxPower: I removed the tT, yet no dice. |
| 16:41.41 | ManxPower | Yourname`: now start doing what [TK]D-Fender told you to do. |
| 16:41.48 | Yourname` | ManxPower: Fix the modeS? |
| 16:41.54 | ManxPower | and leave the tTr off until you have it foxed. |
| 16:41.57 | ManxPower | fixed too. |
| 16:41.59 | ManxPower | Yourname`: correct. |
| 16:42.08 | Yourname` | ManxPower: What are these modes? :S |
| 16:42.34 | ManxPower | dtmfmode=inbamd|info|rfc2833 Whatever you set it to in Asterisk you must also set the same mode on the SIP phone you are using. |
| 16:42.55 | bkw__ | you mean inband |
| 16:42.59 | *** join/#asterisk threewayone (n=hellowor@222.127.173.145) |
| 16:43.00 | ManxPower | inband only works with ulaw and alaw, of course. info is the old way, rfc2833 is the new way. |
| 16:43.04 | threewayone | hi guys |
| 16:43.05 | *** part/#asterisk EnoCix (n=jsloan@216.207.245.1) |
| 16:43.08 | ManxPower | next time try reading The Good Book |
| 16:43.16 | Yourname` | Ohhhhh dtmfmode.. sorry |
| 16:43.45 | ManxPower | If you only set it on one side, I'll send Qwell's Foot of Justice your way. |
| 16:43.58 | bkw__ | shakes his head |
| 16:44.16 | threewayone | does g729 to g729 eat alot of CPU? |
| 16:44.22 | [TK]D-Fender | Yourname`, and you're wondering why DTMF isn't working? |
| 16:44.25 | ManxPower | bkw__: Yourname is one of the hardest users to support. |
| 16:44.30 | threewayone | compared to ulaw to g729 |
| 16:44.30 | hardwire | threewayone: it eats n*x cpu. |
| 16:44.46 | bkw__ | ManxPower: I find him rather easy to work with. |
| 16:44.48 | [TK]D-Fender | threewayone, generally none, jsut like every other like-codec scenario |
| 16:44.57 | ManxPower | bkw__: thanks for volunteering 8-) |
| 16:45.17 | bkw__ | ManxPower: I help him with FreeSWITCH |
| 16:45.29 | [TK]D-Fender | bkw__, He just seems to "need help". |
| 16:45.32 | threewayone | can a core2duo handle 60 simultaneous calls at a time? |
| 16:45.38 | bkw__ | ManxPower: but i'll do my part to help him with Asterisk too |
| 16:45.47 | bkw__ | threewayone: it should have no problem doing that |
| 16:45.59 | Yourname` | What's wrong with needing help? |
| 16:46.02 | threewayone | even with g729 transcoding |
| 16:46.17 | bkw__ | threewayone: hrm it might.. just have to test it for yourself |
| 16:46.18 | ManxPower | threewayone: Can a car tow a boat? As you can see unless you know what kind of car and what kind of boat the question cannot be answered. |
| 16:46.42 | bkw__ | threewayone: if you're doing g729 to g729 it shouldn't use CPU |
| 16:47.03 | Yourname` | It's like you guys have a channel for helping but the best you can do is say "Go do this |
| 16:47.07 | Yourname` | " or "go do that" |
| 16:47.13 | threewayone | ok.. heres my setup: Core2Duo 1.8 GHZ 2GB RAM 2x 300GB SATA call recording to wav |
| 16:47.18 | ManxPower | Yourname`: that is what we do here. |
| 16:47.19 | bkw__ | Yourname`: its been said this channel isn't for support |
| 16:47.21 | Yourname` | If everyone wanted to read the book, or had to.. there won't be a need for this channel. |
| 16:47.45 | bkw__ | threewayone: call recording will nail you with g729 |
| 16:47.48 | ManxPower | Yourname`: There would be a need for this channel -- all the stupid questions would go away. |
| 16:47.59 | Yourname` | bkw__: It's listed as community support http://www.asterisk.org/community |
| 16:48.11 | threewayone | <bkw__> threewayone: call recording will nail you with g729 --> what do you mean? |
| 16:48.12 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 16:48.20 | Assid | heya |
| 16:48.21 | ManxPower | Instead, you want us to spend our free time helping someone that does not want to help themselves. I just find that incredibly rude and annoying. |
| 16:48.22 | bkw__ | threewayone: it'll chew CPU and Disk IO |
| 16:48.30 | Yourname` | ManxPower: Sure. :) |
| 16:48.34 | *** join/#asterisk MaartenB (n=Maarten@84-105-197-29.cable.quicknet.nl) |
| 16:48.45 | bkw__ | ManxPower: but unless you help them how can they help others? I find it easier to help someone then they return the favor in kind. |
| 16:48.47 | [TK]D-Fender | Yourname`, wondering why your DTMF isn't working and lookin at random unrelated crap isn't too bright. Not looking at the DTMFMODE of your phone is also not bright. Especially after we tell you 2-3 times consecutively. |
| 16:48.53 | threewayone | <bkw__> threewayone: it'll chew CPU and Disk IO --> so whats your suggestion? |
| 16:49.05 | ManxPower | bkw__: they could start by reading the damn book |
| 16:49.10 | bkw__ | threewayone: well the woodcrest or clover town chips work great |
| 16:49.19 | bkw__ | ManxPower: some people don't learn that way. |
| 16:49.38 | bkw__ | threewayone: but nobody here can really give you solid numbers on that .. too many variables |
| 16:49.44 | ManxPower | [TK]D-Fender's statement is why Yourname is hard to support. |
| 16:49.49 | Yourname` | [TK]D-Fender: YHou keep saying modes. Bloody modes. |
| 16:50.01 | *** join/#asterisk jasonwoot (n=jasonrot@user-69-73-40-171.knology.net) |
| 16:50.02 | Yourname` | You typed it 2-3 times. |
| 16:50.08 | Nugget | swanky modes! |
| 16:50.10 | ManxPower | [TK]D-Fender: both of us told you what to look at. Now go do it. |
| 16:50.11 | threewayone | ok.. anyway my main goal is just to record calls that go out from another tradional pbx behind asterisk |
| 16:50.17 | cpm | swanky modes! |
| 16:50.21 | ManxPower | threewayone: why are you using G729? |
| 16:50.40 | bkw__ | Yourname`: let me clarify.. you're having DTMF issues .. you'll need to set the DTMF mode for the sip peers so its correct. Their are three modes.. inband, info and rfc2833 (rtp). |
| 16:50.53 | Yourname` | BECAUSE I didn't know what "modes" are. Because you always say things like "dont confuse between extensions and phones" -> I'd think you want to be on the terms to use train at all times. So rather than saying dtmfmode, you had to repeat "modes" like 3 times. |
| 16:50.57 | bkw__ | Yourname`: I recommend rfc2833 |
| 16:51.15 | Assid | hrmm i got recommended of the snom 300.. is that good? i need a < $100 phone |
| 16:51.17 | MaartenB | hello everyone |
| 16:51.18 | bkw__ | Yourname`: if you look in sip.conf and add dtmfmode=rfc2833 to the sip user/peer/friend it'll help |
| 16:51.32 | Yourname` | bkw__: Thanks. I found what [TK]D-Fender was talkiing about after he said dtmfmode. And then changed around and it's fine now. |
| 16:51.37 | mort_gib | Assid: They are fairly decent |
| 16:51.39 | threewayone | <ManxPower> threewayone: why are you using G729? --> from asterisk pbx to voip provider |
| 16:51.41 | ManxPower | well, now three people have told you that, Yourname` Are you going to go try it now. |
| 16:51.46 | MaartenB | when I try to transfer a call to another sip phone (with xfer), the call gets dropped, any suggestions how to fix that? |
| 16:51.49 | [TK]D-Fender | Yourname`, And you can't use some tiny bit of IQ and look at dtmfmode? You instead go on thinking the dialplan controls your phone's inability to signal *? |
| 16:52.15 | bkw_ | OMG why do you all have to act like this? |
| 16:52.18 | Yourname` | [TK]D-Fender: Because it didn't occur to me, as on the CLI I see it reading the DTMF. |
| 16:52.20 | ManxPower | threewayone: that might be a good reason. GSM will reduce your CPU usage drastically with some decrease in call sound quality. |
| 16:52.22 | bkw_ | it turns my stomach to see this |
| 16:52.36 | cpm | continues to scream |
| 16:52.50 | *** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net) |
| 16:52.50 | *** mode/#asterisk [+o mog] by ChanServ |
| 16:52.53 | [TK]D-Fender | Yourname`, jsut because it reads DTMF from one side doesn't mean it can pass it on when the OTHER SIDE sin't set right. |
| 16:53.04 | ManxPower | bkw_: because he does not listen. |
| 16:53.04 | Yourname` | [TK]D-Fender: Hence why I said it didn't occur to me. |
| 16:53.16 | ManxPower | Yourname`: 10 mins ago I told you about dtmfm0ode |
| 16:53.28 | threewayone | <ManxPower> threewayone: that might be a good reason. GSM will reduce your CPU usage drastically with some decrease in call sound quality. -> so ill use gsm instead of g729 |
| 16:53.30 | bkw_ | ManxPower: and did you not see that he has it working already. |
| 16:53.42 | ManxPower | Yourname`: ManxPower: What are these modes? :S |
| 16:53.42 | ManxPower | ManxPower: dtmfmode=inbamd|info|rfc2833 Whatever you set it to in Asterisk you must also set the same mode on the SIP phone you are using. |
| 16:53.53 | Yourname` | No ManxPower, 10 mins ago you said this -> [12:29]<ManxPower>Yourname`: the rtT could be causing that |
| 16:53.56 | ManxPower | then bkw pointed out my speeling error for "inband" |
| 16:54.06 | [TK]D-Fender | Yourname`, .... its 12:52 now |
| 16:54.13 | [TK]D-Fender | Yourname`, sorry, 12:54 |
| 16:54.16 | Yourname` | [12:32]<Yourname`>ManxPower: tT is for transferring, not DTMF. I understand.. but somehow I thought it's connected, but oh well. |
| 16:54.21 | [TK]D-Fender | Yourname`, your sense of time is pretty far off. |
| 16:54.32 | mort_gib | TK: Actually it's 18:54 |
| 16:54.36 | bkw_ | I say be nicer to newbies |
| 16:54.42 | [TK]D-Fender | mort_gib, OUR TIME ZONE. |
| 16:55.13 | mort_gib | TK: Our time zone.... Internet time |
| 16:55.18 | Yourname` | bkw_: Actually, back in the day [TK]D-Fender used to be nice. |
| 16:55.23 | ManxPower | bkw_: he's not a newb, he's been here for months and months |
| 16:55.36 | bkw_ | ManxPower: some people just don't pick up stuff as fast as others. |
| 16:55.40 | Yourname` | I think the whole helping thing gets to you if you happen to see the same person over and over again. |
| 16:56.05 | bkw_ | ManxPower: I find its best to help someone with a problem then force them to return the favor the next time someone else needs help with the same problem :P |
| 16:56.08 | bkw_ | pay it forward |
| 16:56.09 | Yourname` | bkw_: I appreciate it.. but don't worry about it. :) |
| 16:56.22 | [TK]D-Fender | bkw_, his case is dangerously cronic. |
| 16:56.31 | [TK]D-Fender | chronic* |
| 16:56.46 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:57.03 | ManxPower | Yourname`: glad to see your problem is fixed. |
| 16:57.33 | mackes-Office | Assid: Pick up a Polycom 320. You will be happier in the long run |
| 16:57.44 | bkw_ | 320's don't do NAT |
| 16:57.45 | Assid | how much is the 320 |
| 16:57.52 | Assid | 301's do |
| 16:58.03 | Assid | atleast im using it |
| 16:58.54 | mackes-Office | About 85 |
| 16:59.03 | Assid | is 301> 320 ? |
| 16:59.09 | [TK]D-Fender | bkw_, what do you mean "don't do NAT"> |
| 16:59.11 | mackes-Office | Polycom 320: http://www.ipphone-warehouse.com/Polycom-Soundpoint-IP-320-2200-12320-025-p/2200-12320-025.htm |
| 16:59.27 | [TK]D-Fender | bkw_, something specific, because I've run Polycom's behind NAT before just fine. |
| 16:59.31 | bkw_ | no STUN |
| 16:59.34 | bkw_ | the proper way to do nat is STUN |
| 16:59.50 | [TK]D-Fender | Assid, 320/330 > 301 |
| 17:00.02 | ManxPower | I was not aware that NAT and STUN were the same thing. |
| 17:00.04 | mort_gib | Quick question, I need to change the default MOH, do I have to install mpeg123?? Or is it better to prepare the mp3 files?? |
| 17:00.09 | Assid | so how come the 320 cheaper? |
| 17:00.27 | [TK]D-Fender | Assid, because they sell a LOT of these. its less plastic as well. |
| 17:00.28 | mackes-Office | Then what? |
| 17:00.28 | ManxPower | mort_gib: My recommendation is to not use mp3 files at all. |
| 17:00.33 | [TK]D-Fender | Assid, the 301 is EOL |
| 17:00.53 | anonymouz666 | ManxPower: indeed. I got a crash today with format_mp3. |
| 17:00.58 | redback | mort_gib: the system would have to decode/encode the mp3 each time which can be quite CPU intensive |
| 17:01.05 | ManxPower | mort_gib: as of 1.4 asterisk supports MOH in any format asterisk supports with no mpg123 or format_mp3 required. |
| 17:01.08 | Assid | okay .. these guys need a new one. so 330 ? |
| 17:01.42 | [TK]D-Fender | ManxPower, umm... thats a little circular. |
| 17:01.45 | mackes-Office | Really? MP3 is supported natively for MOH in 1.4? That rocks! |
| 17:01.55 | [TK]D-Fender | mackes-Office, he didnt' acktually say that. |
| 17:01.56 | bkw_ | ManxPower: they aren't.. the proper way to do SIP from behind nat is STUN in most cases. |
| 17:02.15 | mackes-Office | oh. Bummer |
| 17:02.18 | bkw_ | ManxPower: you're so nice about things.. keep up the good work. |
| 17:02.22 | [TK]D-Fender | bkw_, I've never used STUN on any device for NAT scenarios before... |
| 17:02.24 | ManxPower | mackes-Office: No it is not. |
| 17:02.30 | bkw_ | [TK]D-Fender: I have.. works great |
| 17:02.35 | Assid | bkw_: hrmm never needed stun on my 301 |
| 17:02.41 | Assid | works fine |
| 17:02.47 | bkw_ | [TK]D-Fender: its just one of many tools in the fight against NAT :P |
| 17:02.55 | bkw_ | or with NAT I should say |
| 17:02.59 | mort_gib | I don't much care, what format I use, I would prefer to encode correctly first actually |
| 17:03.13 | mackes-Office | So, 1.4 will play mp3's for MOH or now? |
| 17:03.16 | [TK]D-Fender | bkw_, I'm sure it is, but generally doesn't seem necessary.. that just helps the phone know better how to deal with thing, and well.. in general it just hasn't seemed to mater... |
| 17:03.21 | mackes-Office | not? |
| 17:03.25 | mort_gib | -So I'm better off using wav files?? |
| 17:03.28 | [TK]D-Fender | mackes-Office, install format-mp3.so <- |
| 17:03.35 | mackes-Office | Oh, ok |
| 17:03.37 | [TK]D-Fender | mort_gib, same for you |
| 17:03.37 | mackes-Office | I see |
| 17:03.46 | *** part/#asterisk bkw_ (n=brian@adsl-71-153-169-69.dsl.tul2ok.sbcglobal.net) |
| 17:03.49 | Assid | okay between 320 and 330 .. what would better |
| 17:03.51 | mort_gib | Yes, but I'm worried about performance :-) |
| 17:04.00 | [TK]D-Fender | Assid, 330 has a pass-through port. Only difference |
| 17:04.09 | [TK]D-Fender | mort_gib, then convert them |
| 17:04.31 | sp00kz | having the sound format in its native compression is always best, less problem for servers, it doesnt need to be reencoded to send to the phone/outside |
| 17:04.46 | mort_gib | Okay... |
| 17:06.39 | *** join/#asterisk vector (n=vector@host-178-246-220-24.midco.net) |
| 17:06.42 | *** join/#asterisk shido6 (n=shido6@74-130-120-3.dhcp.insightbb.com) |
| 17:07.13 | ManxPower | Looks like the IT director screwed up DNS again. He forgot to put in an MX record for the company's primary domain. And we all know what problem that will cause. |
| 17:08.07 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 17:09.29 | threewayone | if your voip provider supports g729 will it lessen cpu load |
| 17:10.38 | Assid | hrmm |
| 17:10.51 | Assid | im thinking of using g729.. problem is the licensing if i want to put it to voicemail |
| 17:11.21 | threewayone | k.. anyway im more concerned with outbound calls using g729 |
| 17:11.25 | ManxPower | Assid: you will need G729 for voicemail, call recording, DTMF Transfer Hack, IVRs, etc |
| 17:11.38 | Assid | hrmm too many issues |
| 17:11.52 | ManxPower | Assid: so it's much easier to just buy the required licenses |
| 17:12.48 | threewayone | ManxPower: If I record those outbound calls what file format is best to save cpu load |
| 17:12.58 | [TK]D-Fender | Assid, No issue for voicemail. |
| 17:13.20 | Assid | but then i gotta save it back in 729 |
| 17:13.25 | [TK]D-Fender | ManxPower, MMETME however will suck tremendously ;) |
| 17:13.29 | [TK]D-Fender | MEETME* |
| 17:13.34 | Assid | hrmm |
| 17:13.49 | [TK]D-Fender | Assid, and the problem with saving VM's as G.729? |
| 17:14.06 | Assid | hrmm need codec :P |
| 17:15.19 | [TK]D-Fender | Assid, ....? |
| 17:15.42 | Assid | trying to save some codec licensing costs |
| 17:15.44 | [TK]D-Fender | Assid, if EVERYTHING is done in G.729 you don't need the codec. |
| 17:15.49 | Assid | oh |
| 17:15.50 | [TK]D-Fender | Assid, pay attention. |
| 17:16.03 | [TK]D-Fender | Assid, you need to pay to TRANSCODE to/from G.729 |
| 17:16.47 | Assid | hrmm k.. i thought even for voicemail since it saves a 729 stream |
| 17:17.22 | *** join/#asterisk AndyGraybeal (n=AndyGray@128-177-27-78.ip.openhosting.com) |
| 17:17.29 | [TK]D-Fender | Assid, no cost to saving a strem, only to encode/decode. |
| 17:17.40 | [TK]D-Fender | Assid, if it gets written as-is then its just packets. |
| 17:17.50 | Assid | k |
| 17:19.22 | *** join/#asterisk jtexter3 (n=jamest@adsl-154-42-229.asm.bellsouth.net) |
| 17:21.22 | threewayone | which is better to link 2 asterisk boxes, sip or iax? |
| 17:21.59 | threewayone | asterisk1 just forwards calls to asterisk2 which forwards calls to the voip provider |
| 17:22.06 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 17:22.06 | threewayone | via sip |
| 17:22.13 | hardwire | threewayone: use tdmoe for ultimate pleasure. |
| 17:22.16 | jtexter3 | I'm having an issue with one way audio. I have Polycom IP 550's, and a Digium AEX800 with a 4 port FXO module. Audio from extension to extension is fine, but on the PSTN, I get one way audio. I've removed all features so it's a simple Dial(Zap/g1/${EXTEN:1}. I don't see anything unusual in the logs. Any thoughts? This is asterisk 1.4.19.1 and zaptel 1.4.10.1 |
| 17:22.42 | hardwire | threewayone: actually.. iax2 is awesome, read up on iax2 trunking. |
| 17:23.05 | [TK]D-Fender | threewayone, If you can spare the bandwidth, stay with SIP |
| 17:23.09 | Qwell | jtexter3: have you confirmed that audio is getting to Asterisk from the polycom? |
| 17:23.16 | Qwell | If so, that seems quite odd. I would recommend contacting Digium support |
| 17:23.46 | jtexter3 | Qwell: I'll verify, but I believe so |
| 17:23.50 | threewayone | asterisk1 and astersik2 are on the same box |
| 17:23.54 | Qwell | jtexter3: rtp set debug on |
| 17:24.08 | Qwell | in which direction is the audio not flowing? |
| 17:24.11 | threewayone | asterisk1 and astersik2 are on the same LAN rather sorry |
| 17:24.25 | Qwell | to the phone, or from the phone? the latter is easy to verify in logs |
| 17:25.14 | [TK]D-Fender | jtexter3, I'd suggest independently testing each end with Record and Playback. |
| 17:25.48 | threewayone | <[TK]D-Fender> threewayone, If you can spare the bandwidth, stay with SIP --> asterisk1 and asterisk2 are on the same LAN, asterisk1 is a predictive dialer which passes calls to asterisk2, asterisk2 passes it to the voice provider and records the call |
| 17:26.02 | [TK]D-Fender | threewayone, SIP it is. |
| 17:26.37 | threewayone | <[TK]D-Fender> threewayone, SIP it is. --> Thanks because I tried using iax trunk and i got choppy lines |
| 17:26.50 | jtexter3 | The party on the PSTN can hear, but the person on the Polycom cannot |
| 17:27.05 | jtexter3 | so, traffic to the phone |
| 17:27.11 | Qwell | that's a little more difficult to verify |
| 17:27.39 | *** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net) |
| 17:27.47 | Qwell | is it just the one phone that has problems, or all calls through the PSTN? Can you try testing a softphone on the same box or anything like that? |
| 17:27.52 | *** join/#asterisk adr3nalin3 (n=afink@asa.redglaze.com) |
| 17:28.23 | Bananaskin | Hey guys, any reason why modules don't seem to compile under 1.4.18.1? |
| 17:28.51 | Qwell | Bananaskin: gonna need to be more specific. can you pastebin the Make errors? |
| 17:28.52 | Qwell | ~pb |
| 17:28.53 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 17:30.19 | Bananaskin | Qwell, sorry, there are no errors that I can see, it appears not to compile the modules in the /apps dir. |
| 17:31.21 | jtexter3 | Qwell: All calls to the PSTN. Unfortunately, I'm remote, but I'll see if I can get a softphone going locally and try that out |
| 17:31.46 | Qwell | Bananaskin: are they enabled in menuselect? |
| 17:32.01 | Qwell | things like app_meetme require zaptel to be installed |
| 17:32.11 | Bananaskin | yep, all enabled, and the modules dir is cleared before the make and make install |
| 17:32.24 | Qwell | which ones, specifically, aren't being built/installed? |
| 17:32.27 | Bananaskin | all |
| 17:32.41 | Qwell | O.o |
| 17:32.46 | Bananaskin | I am having to fudge a fix in app_voicemail |
| 17:33.02 | Qwell | how, exactly, are you building? start from downloading/unpacking the source |
| 17:33.08 | Bananaskin | wrong perms are being applied to the vm txt file and rendering the vm useless |
| 17:33.09 | Qwell | what steps are being used |
| 17:33.31 | Bananaskin | for the compile ? |
| 17:33.41 | Qwell | yes |
| 17:33.56 | Bananaskin | ./configure, make menuselect, make, and make install |
| 17:33.57 | Qwell | please don't skip any steps, even if you think it might be trivial |
| 17:34.04 | Qwell | what are you doing in make menuselect? |
| 17:34.31 | Bananaskin | enabling the modules and audio files/moh etc |
| 17:35.16 | *** join/#asterisk horvath (n=zzz@bas1-toronto26-1279484350.dsl.bell.ca) |
| 17:35.49 | Bananaskin | looking back through, I see that there references to the modules being compiled.. |
| 17:35.51 | horvath | How can I completely disable T38 on my asterisk 1.4x box? I keep getting Unsupported SDP media type in offer etc etc |
| 17:35.57 | Qwell | Bananaskin: but not installed? |
| 17:36.18 | Bananaskin | no .so are created in the app dir either |
| 17:36.38 | Qwell | can you pastebin your menuselect.makeopts file? |
| 17:36.43 | Bananaskin | kinda confusing :) |
| 17:36.46 | Bananaskin | sure |
| 17:37.12 | hi365 | are there any dial args that need to be passed for applicationmap to work? |
| 17:37.19 | hi365 | (features.conf) |
| 17:37.22 | Fusoya | Argh... does anyone know of an easy way that I can make Asterisk 1.2 log or send an event every time a call hangs up? |
| 17:37.38 | hi365 | Fusoya: do somethign in the h exten |
| 17:37.52 | hi365 | (of your call context) |
| 17:38.10 | Fusoya | hi365: Hmmm OK |
| 17:38.38 | threewayone | un |
| 17:38.57 | Fusoya | I won't be able to determine the extension or the call ID with the h extension, though? |
| 17:39.22 | [TK]D-Fender | Fusoya, what do you actualyl want to do? |
| 17:40.37 | Bananaskin | Qwell, - http://pastebin.ca/1022936 |
| 17:40.48 | [TK]D-Fender | hi365, one of tTwW |
| 17:40.59 | Qwell | Bananaskin: you have module embedding enabled |
| 17:41.04 | Fusoya | [TK]D-Fender: I have a recorder connected trunk-side to the T1s going into the Asterisk box |
| 17:41.09 | Qwell | This is precisely what that option does. |
| 17:41.28 | [TK]D-Fender | Fusoya, Ok, and....? |
| 17:41.29 | Fusoya | I need to send events to that recorder to tell it when the calls end... ideally, I need to send several pieces of information, including the channel, the extension, and the asterisk call ID |
| 17:41.29 | Bananaskin | ahhh :) well that clears that up, cheers for that |
| 17:41.29 | hi365 | [TK]D-Fender: any of them will activate the applicationmap features as well? |
| 17:42.03 | [TK]D-Fender | hi365, should. |
| 17:42.10 | hi365 | thanks |
| 17:42.22 | hi365 | did. thanks |
| 17:42.32 | Fusoya | I can't figure out an easy way to make asterisk 1.2 generate those events |
| 17:42.43 | [TK]D-Fender | Fusoya, what is this "recorder"? |
| 17:42.50 | Fusoya | [TK]D-Fender: Familiar with Tantacomm? |
| 17:43.10 | Fusoya | It's a Tantacomm auditor. |
| 17:43.52 | [TK]D-Fender | Fusoya, Guess you need that being 3rd party. What I might suggest is to use SQL for CDR storage and write a stored procedure on write. |
| 17:44.36 | Fusoya | [TK]D-Fender: That would definitely be a more robust solution, and sounds like the way to go, but unfortunately would require a lot of restructuring... |
| 17:44.45 | Fusoya | Naturally, I'm looking for something Q&D |
| 17:45.11 | [TK]D-Fender | Fusoya, Or perhaps you could monitor AMI for the channel close signal. |
| 17:46.06 | Fusoya | Hmmm |
| 17:46.27 | Fusoya | I may end up just dropping in the CTI server that Tantacomm is trying to push on us |
| 17:47.32 | hi365 | hmm, "The applicationmap is not intended to be used for all Asterisk applications....Examples of this would be things like Macro..." |
| 17:47.32 | hi365 | ( http://svn.digium.com/view/asterisk/branches/1.4/configs/features.conf.sample?view=markup ) |
| 17:47.32 | hi365 | So i cant run a macaro. Is there any way to do validation ( I would like to promtp the user for an extension to transfer the call, but I need to do some validation on the dest. exten.)? |
| 17:47.33 | *** join/#asterisk horvath (n=horvath@bas1-toronto26-1279484350.dsl.bell.ca) |
| 17:48.21 | [TK]D-Fender | hi365, this is sounding unnecessarily complex. What are you actually trying to accomplish? |
| 17:48.40 | [TK]D-Fender | Fusoya, So you don't have this unit yet? |
| 17:48.56 | Fusoya | No, we're flying without a CTI server at the moment |
| 17:48.56 | hi365 | ( I would like to promtp the user for an extension to transfer the call, but I need to do some validation on the dest. exten. i.e. only alow SOME exten's to transfer to SOME exten's)? |
| 17:49.09 | horvath | So guys... any idea how I can completely disable t38? I dont want asterisk even trying to negotiate t38 |
| 17:49.20 | [TK]D-Fender | Fusoya, So you're pure Avaya right now? |
| 17:49.35 | Fusoya | SER |
| 17:49.54 | [TK]D-Fender | Fusoya, Ah. think I got that other bit from Googling Tantacomm |
| 17:50.05 | [TK]D-Fender | Fusoya, So waht are you trying to do? |
| 17:50.08 | file | horvath: usually the remote side sends a T38 reinvite if it wants to talk T38 to Asterisk... so you'd have to disable it on the device |
| 17:50.32 | Fusoya | We need to be able to use this tantacomm box to send recordings of inbound calls we take to our client |
| 17:51.07 | Fusoya | The problem is that we need a way to be sure that there's no audio bleed-over on those recordings from subsequent calls on the same channel |
| 17:51.19 | [TK]D-Fender | Fusoya, * can already record calls. What part or *'s capabilities are insufficient? |
| 17:51.23 | Katty | hai. |
| 17:51.27 | horvath | file: My incoming sip provider is trying to do t38 but I want asterisk to just reject t38 and use ulaw without giving me a ton of error messages like Unknown RTP codec 102 received from |
| 17:51.36 | Fusoya | [TK]D-Fender: Absolutely nothing. It's a boneheaded contractual thing. |
| 17:51.46 | hi365 | [TK]D-Fender: I would like to promtp the user for an extension to transfer the call, but I need to do some validation on the dest. exten. i.e. only alow SOME exten's to transfer to SOME exten's |
| 17:51.48 | file | horvath: if you don't enable it, it does reject it |
| 17:51.58 | [TK]D-Fender | Fusoya, that sums it up. |
| 17:52.08 | horvath | file: and yet it throws up a ton of Unknown RTP codec 102 received from msgs |
| 17:52.13 | *** join/#asterisk bkruse (n=bkruse@216.207.245.1) |
| 17:52.13 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 17:52.21 | *** join/#asterisk jmls_net (n=asterisk@host217-36-208-155.in-addr.btopenworld.com) |
| 17:52.21 | [TK]D-Fender | horvath, And how do those "messages" really affect your life? |
| 17:52.26 | jmls_net | evening all |
| 17:52.34 | [TK]D-Fender | Katty, Mew. |
| 17:52.34 | Fusoya | [TK]D-Fender: Actually, we already record the calls *twice*... using asterisk and using a SAaS vendor. Not good enough. :) |
| 17:52.59 | jmls_net | using the latest 1.4 svn - is there anyone who is using ChanSpy in anger ? My previous experiences have not been that, um, good ... |
| 17:53.00 | Fusoya | Anyway, I'll keep plugging away. Thanks for your help! |
| 17:53.00 | [TK]D-Fender | Fusoya, putting the "anal" in anal-retentive/. |
| 17:53.05 | horvath | [TK]D-Fender: Yes... I feel like aliens inside asterisk are trying to communicate with me |
| 17:53.21 | Fusoya | [TK]D-Fender: something like that |
| 17:53.32 | [TK]D-Fender | horvath, Oh, and here I thought it was actually important... |
| 17:54.06 | file | runs off to a meeting |
| 17:55.38 | horvath | [TK]D-Fender: I guess your right.. as they are just notices not errors |
| 17:56.39 | horvath | [TK]D-Fender: I'm still having an issue after a fax is recieved sucessfully and everything prints out fine the other end gets a line error as if the call wasen't hung-up or the fax didn't go though (which it did) |
| 17:57.18 | [TK]D-Fender | horvath, faxing over VoIP? Not a good idea. |
| 17:57.48 | horvath | [TK]D-Fender: Yes but its on a LAN basically |
| 17:57.51 | hardwire | [TK]D-Fender: use hylafax + iaxmodem (hylafax client for remote faxing) |
| 17:58.02 | hardwire | horvath: it's still not great. |
| 17:58.33 | *** join/#asterisk CaTtleyA (n=DDoS@bas1-montrealak-1128581558.dsl.bell.ca) |
| 17:58.48 | horvath | Really? Even with no latency |
| 17:59.06 | [TK]D-Fender | hardwire, I just set that up at home here as a test. What do you use to manage Hylafax? |
| 17:59.19 | hardwire | vim |
| 17:59.35 | hardwire | it's pretty multi-purpose. |
| 17:59.45 | dFence | why can't i access the ${CDR(foo)} variables after a call within the h-extension? |
| 17:59.54 | hardwire | punches [TK]D-Fender in the eye. |
| 18:00.27 | *** join/#asterisk Bananaskin (n=mike@user-5444d76a.lns1-c11.dsl.pol.co.uk) |
| 18:00.36 | dFence | i have hyla/iaxmodem running and so far - no complaints |
| 18:00.44 | [TK]D-Fender | hardwire, I meant you don't use any front ends for mass-faxing, etc? |
| 18:00.46 | hardwire | dFence: it's pretty spiffy. |
| 18:00.55 | *** join/#asterisk gardo (n=gardo@121.97.178.31) |
| 18:00.59 | Bananaskin | Qwell, that worked a treat, and the VM problems have gone as well, thanks again |
| 18:01.00 | hardwire | [TK]D-Fender: I've been testing several clients, none of which I really like |
| 18:01.01 | [TK]D-Fender | hardwire, sure inbound is easy enough, I'm looking for outbound/ |
| 18:01.04 | hardwire | then again I don't like faxing. |
| 18:01.07 | hi365 | anyone have a sample of what a transfer context is supposed to look like? |
| 18:01.07 | *** join/#asterisk creativx (n=creadure@226.62-97-205.bkkb.no) |
| 18:01.14 | hardwire | [TK]D-Fender: gfax worked well in linux |
| 18:01.18 | [TK]D-Fender | hardwire, neither do I, its just something I need to do at my company. |
| 18:01.29 | hardwire | I think I used a java one that worked well |
| 18:01.41 | hardwire | it's neat being able to relate in/out faxes to a user id |
| 18:01.54 | hardwire | the client sees faxes in the spool and lets you access old/new ones. |
| 18:05.55 | *** join/#asterisk CVirus (n=GoD@62.135.96.15) |
| 18:06.59 | *** join/#asterisk znoG_ (n=gs@host49.190-139-153.telecom.net.ar) |
| 18:09.11 | horvath | Any experiences with avantfax? |
| 18:09.35 | [TK]D-Fender | horvath, I bookmarked them to look at later. |
| 18:10.37 | *** join/#asterisk angom (n=angom@201.170.65.143) |
| 18:14.37 | jaytee | [TK]D-Fender, you know they have this thing called sleep? I hear it's really good for your health! :-) |
| 18:15.06 | [TK]D-Fender | jaytee, its 2:14pm... why would I be asleep? |
| 18:15.20 | creativx | so you could be awake... in the night |
| 18:15.48 | dFence | ok.. something's definitely not right... |
| 18:15.50 | jaytee | [well, you were still up and chatting at around 3am this morning |
| 18:16.31 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 18:16.34 | [TK]D-Fender | jaytee, firs t time I've been up that late in a long time. Holiday today. |
| 18:17.02 | jaytee | yeah, I did two days back to back like that this weekend working on mysql stuff |
| 18:17.33 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 18:17.43 | jaytee | [TK]D-Fender, by Holiday you mean what we "Yanks" call a vacation day, right? |
| 18:18.16 | hi365 | is this line a valid transfer-extensions? exten => _2xx,1,Dial(Local/${EXTEN}@from-internal) |
| 18:18.44 | [TK]D-Fender | jaytee, No. Vacation is when you use days you are entitled to. Holiday is a fixed "business stops" day like Thanksgiving / Christmas, etc |
| 18:19.08 | jaytee | so what's the Holiday in Canada today? |
| 18:19.19 | [TK]D-Fender | hi365, don't call that a "transfer"/ |
| 18:19.48 | hi365 | [TK]D-Fender: what is then? does it need to be a goto? |
| 18:19.48 | [TK]D-Fender | jaytee, "Victoria Day" (Canada at large), "Fete du Dollard" (Quebec) |
| 18:20.00 | [TK]D-Fender | hi365, Are you looking to do this for a single exten? |
| 18:20.17 | hi365 | from any extension to a range |
| 18:20.27 | [TK]D-Fender | hi365, what you are doing looks like ti deserves to be an "include =>" |
| 18:20.39 | hi365 | hmm |
| 18:21.16 | hi365 | but i would like to limit it to the range... wont the include include every thing in the included context? |
| 18:22.44 | [TK]D-Fender | perhaps you should break up the context you want to link to. |
| 18:23.50 | hi365 | is that the only way? does that mean that transfer-context (if its not the actual context) is just a "link"? |
| 18:24.08 | [TK]D-Fender | hi365, HUH? |
| 18:24.18 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 18:24.43 | hi365 | ok, lets start from the begining: How do i build a transfer-context? |
| 18:24.57 | [TK]D-Fender | hi365, What is "transfer-context"? |
| 18:25.21 | jameswf-home | I would build a transfer macro |
| 18:25.26 | hi365 | the context used by ## (or whatever you have set in features.conf) |
| 18:26.54 | [TK]D-Fender | hi365, that term does not exist really. |
| 18:26.58 | hi365 | jameswf-home: with what, dial statments? |
| 18:27.25 | jameswf-home | depends what you want it to do... |
| 18:27.33 | hi365 | is reading form here: http://www.voip-info.org/wiki/view/Asterisk+config+features.conf |
| 18:27.48 | hi365 | "If you set the variable __TRANSFER_CONTEXT, then that context will be used...." |
| 18:27.54 | [TK]D-Fender | hi365, separate the extens in [from-internal] into a separate context. Have [from-interal] include it. T hen include it in the context you though of shoving that pattern match into. |
| 18:28.53 | [TK]D-Fender | hi365, Why is it you're using DTMF for transfers? |
| 18:29.11 | hi365 | how else? |
| 18:30.09 | hi365 | if the "transfer-context" can be ANY context then then why wouldnt this work? exten => _2xx,1,Dial(Local/${EXTEN}@from-internal) |
| 18:30.49 | *** part/#asterisk horvath (n=horvath@bas1-toronto26-1279484350.dsl.bell.ca) |
| 18:31.34 | [TK]D-Fender | hi365, have you tried a Goto? |
| 18:31.59 | [TK]D-Fender | hi365, And that will work, its just really ugly, and will spam CDR's and creat unnecessary channels |
| 18:32.08 | hi365 | yes, but ill try again |
| 18:32.32 | hi365 | actualy, the dial sent my system load to 415.2 |
| 18:32.43 | *** join/#asterisk skirmisha (n=viki@79-100-60-165.btc-net.bg) |
| 18:33.01 | skirmisha | guys in 1.4 ver can host take masks as well? |
| 18:33.03 | hi365 | some sort of endless loop |
| 18:33.19 | skirmisha | like 192.168.0.1/24 |
| 18:34.15 | skirmisha | ??? |
| 18:34.16 | hi365 | yup - dial leads to an endless loop |
| 18:34.49 | dFence | i have a exten => _0.,1,DIAL(CAPI/contr1/${EXTEN:1}) -- when I dial it from a sip-phone i get a nasty *beep* person-calling-not-available-yadayada... when i use the CLI and DIAL 086@r-out it works just fine - WHY!? |
| 18:35.17 | ManxPower | dFence: the only reason would be if you did it in your dialplan. |
| 18:35.27 | ManxPower | pastebin the complete _0. extension |
| 18:35.52 | [TK]D-Fender | dFence, pastebin both of these |
| 18:36.19 | dFence | http://pastebin.com/m100152b2 |
| 18:37.20 | ManxPower | dFence: ALL extensions MUST start with a priority 1 |
| 18:37.22 | *** join/#asterisk uluatu (n=deg@200.195.161.164) |
| 18:37.40 | dFence | whops, forgot to change that |
| 18:37.44 | ManxPower | Nothing in that dialplan would play person calling not available |
| 18:37.55 | dFence | ManxPower: well.. kinda does ;D |
| 18:38.16 | ManxPower | dFence: where is the Playback(person-calling-not-available) |
| 18:38.29 | ManxPower | Asterisk does not play sounds unless you tell it to. |
| 18:38.38 | dFence | dFence: auto fallthrough i guess, hang on - gonna attache the verbose-msg |
| 18:38.50 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 18:38.58 | *** join/#asterisk RoyK (n=roy@ti0002a380-0015.bb.online.no) |
| 18:39.07 | ManxPower | dFence: even with auto fallthru I don't see the playback. |
| 18:39.26 | dFence | http://pastebin.com/d318605a8 |
| 18:39.43 | dFence | oh... |
| 18:40.01 | dFence | just saw that capi info |
| 18:40.06 | *** join/#asterisk mackes-Office (n=root@mail.deltasoniccarwash.com) |
| 18:40.07 | ManxPower | dFence: So you are NOT getting an audio message |
| 18:40.26 | dFence | ManxPower: *darrn* forgot that xlite plays the audio-message itself |
| 18:40.52 | *** join/#asterisk rgsteele||work (n=chatzill@74.94.57.81) |
| 18:42.24 | *** part/#asterisk iamhrh (n=iamhrh@74.7.128.162) |
| 18:42.46 | ManxPower | looks over his glasses at dFence |
| 18:42.48 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 18:43.11 | dFence | well.. doesn't change the fact that calls via CAPI fail when executed by the SIP-Phone |
| 18:43.36 | ManxPower | no it doesn't. |
| 18:43.42 | ManxPower | you are using the r-out context when dialing from the CLI |
| 18:43.55 | ManxPower | try using the Zimmer context |
| 18:44.18 | rgsteele||work | Hey folks. I'm interested in setting up an Asterisk PBX. Do I need anything other than a data line? Or, do I need to notify the telco that I'm interested in doing VOIP, so I can talk via h323 (or some other protocol) to the telco's VOIP equipment? |
| 18:44.23 | dFence | ManxPower: no, Zimmer as well |
| 18:44.34 | ManxPower | dFence: show us |
| 18:44.52 | ManxPower | rgsteele||work: start by reading The Good Book |
| 18:44.53 | ManxPower | ~book |
| 18:44.54 | 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 --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 18:45.24 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 18:45.31 | dFence | http://pastebin.com/d3bed822c |
| 18:45.35 | ManxPower | dFence: to emulate the SIP phone you would need to CLI dial "084@Zimmer" |
| 18:45.46 | dFence | that's what i did |
| 18:46.04 | ManxPower | dFence: I meant show the dial from the CLI |
| 18:46.24 | dFence | ManxPower: starts at line 30 |
| 18:46.29 | maqr | how do i test to see if ztdummy is actually working right? |
| 18:46.36 | ManxPower | BTW, I was not aware that the number "84" is a valid number to dial |
| 18:47.20 | dFence | ManxPower: asterisk is connected to our current pbx's S0-Bus, 84 is the phone in front of me |
| 18:47.23 | ManxPower | dFence: I can see no reason for your problem |
| 18:47.29 | dFence | ManxPower: me neither... |
| 18:47.54 | dFence | i really dunno what "ISDN1#02: CAPI INFO 0x34bf: Service or option not available, unspecified" is supposed to mean... |
| 18:48.10 | ManxPower | dFence: 84 is not the phone you are calling from, is it? |
| 18:49.04 | dFence | ManxPower: no. 84 is the extension of a hardphone at the local pbx |
| 18:52.54 | [TK]D-Fender | maqr, use it |
| 18:53.50 | *** join/#asterisk pikachu2000 (n=pikachu2@pikachu.cyberdesigns.co.za) |
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| 18:58.55 | *** join/#asterisk skirmisha (n=viki@79-100-60-165.btc-net.bg) |
| 18:59.06 | skirmisha | can i use allow=alwa,ulaw or i need 2 rows? |
| 18:59.15 | [TK]D-Fender | 2 |
| 19:00.18 | *** part/#asterisk Ksilebo (n=haxardag@adium/Ksilebo) |
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| 19:03.13 | *** join/#asterisk rvhi (n=chatzill@54.sub-75-209-175.myvzw.com) |
| 19:03.29 | errr | it is possible to use the manager to set the hint of a peer? |
| 19:03.41 | rvhi | hi, after restart *, all hints are screwed up, lots of them showing 'unavailable' now |
| 19:03.45 | rvhi | any idea? |
| 19:03.49 | [TK]D-Fender | errr, to do what? |
| 19:04.18 | errr | [TK]D-Fender: well I want to set the hint to be in use when a user goes into dnd, and I use manager to place them on dnd |
| 19:04.43 | [TK]D-Fender | errr, no viable way to integrate that. |
| 19:04.53 | [TK]D-Fender | errr, how do they go on DND currently? |
| 19:04.54 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 19:05.25 | errr | [TK]D-Fender: they press the DND button which executes a manager script I have that sets a db value for DND |
| 19:06.08 | [TK]D-Fender | errr, problem is you can't modify a non-devstate hint |
| 19:06.19 | *** join/#asterisk mattchis (n=IceChat7@adsl-99-130-234-246.dsl.hstntx.sbcglobal.net) |
| 19:06.45 | *** part/#asterisk jmls (n=asterisk@host217-36-208-155.in-addr.btopenworld.com) |
| 19:07.14 | errr | hmm, my goal is to make it so the recp knows when a user is "busy", right now everything except dnd works for that. Any ideas what I can do to make that work? |
| 19:07.55 | [TK]D-Fender | errr, you could hack a GUI panel together to intpret the state as a sum of 2 different flags like that. |
| 19:08.03 | [TK]D-Fender | errr, but not through a single presence reg |
| 19:08.11 | errr | :( |
| 19:08.20 | mattchis | Does anyone know of a good way to play streams on moh? |
| 19:08.42 | *** join/#asterisk talntid (n=erict@66.208.251.170) |
| 19:08.49 | *** join/#asterisk unbkbl (n=work@static-adsl201-232-88-87.epm.net.co) |
| 19:13.33 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 19:17.03 | deeperror | Every few days I get a port on a channel bank that will just stop dialing. After pressing 1 digit it will go fast busy. After looking around I noticed when doing a zap show channel # that the channel that was dead in the Context: had special characters and garbage in there as if it was overwritten or corrupted in ram. |
| 19:19.41 | tzafrir | memory corruption in Asterisk? |
| 19:19.50 | *** join/#asterisk [hC] (n=hardcore@S01060016b6b53c0c.vc.shawcable.net) |
| 19:19.54 | *** join/#asterisk kerill (n=blades20@85.134.139.148) |
| 19:19.59 | deeperror | thats my guess on it |
| 19:20.17 | kerill | took a while to log on to here |
| 19:20.20 | tzafrir | this is indeed what your description suggests |
| 19:20.38 | deeperror | tzafrir: here is an example http://pastebin.ca/1023050 |
| 19:21.26 | *** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com) |
| 19:21.42 | tzafrir | deeperror, what should the context name have been? |
| 19:21.54 | deeperror | its below custom-set-bank-cid |
| 19:22.16 | deeperror | the entire bank is setup going to the same context |
| 19:22.37 | deeperror | so why only one channel is effected is odd and i'm not exactly sure what caused it to happen but it does occur about every 3-4 days |
| 19:22.57 | deeperror | i've got 50 agents dialing non stop all day so kinda hard to pinpoint the cause but i see why it rings busy now at least |
| 19:23.26 | rgsteele||work | ManxPower: Aafter scanning the relevant chapters (8 and 9), I didn't see anything about what I need outside of the Asterisk box itself, other than a T1 card (as I don't really want to deal with incoming POTS lines) |
| 19:23.39 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 19:25.06 | unbkbl | hello i have a question and it connects and I can make local and outbound calls. Callers however cannot call me whether internal or callers that select my extension from the inbound lines. They go directly to voicemail. Also the phone is showing red under view extensions. |
| 19:25.06 | *** join/#asterisk Aziraphale (n=dfgh@91.142.239.147) |
| 19:26.09 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 19:26.13 | unbkbl | hello i have grandstream telephone with a sip account in a PBX, it connects and I can make local and outbound calls. Callers however cannot call me whether internal or callers that select my extension from the inbound lines. They go directly to voicemail. Also the phone is showing red under view extensions. |
| 19:27.29 | unbkbl | so if some body wants to callme gets a 503 unavaliable error |
| 19:28.05 | deeperror | unbkbl: check router/ports |
| 19:28.35 | unbkbl | and when i seed sip show peer "number of extension" i get an unreachable message |
| 19:31.55 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 19:32.50 | *** join/#asterisk RoyK (n=roy@ti0002a380-0015.bb.online.no) |
| 19:37.27 | Aziraphale | quick question: asterisk server with 2 interfaces, one on the internet, the other a private voice trunk (unreachable) server sends phone an invite with the unreachable address = phone sends rtp stream there - one-way audio. only when the phone is the caller, works ok when called |
| 19:37.38 | *** join/#asterisk CaTtleyA (n=DDoS@bas1-montrealak-1128581558.dsl.bell.ca) |
| 19:37.48 | Aziraphale | anyone come across that before? |
| 19:37.55 | unbkbl | wich port use the phone for establish outbound calls? |
| 19:38.47 | Aziraphale | udp port or server interface? |
| 19:40.45 | RoyK | Aziraphale: I somehow doubt the phone uses the server interface |
| 19:41.18 | Aziraphale | well ,the SIP session does. sip port is 5060 iirc |
| 19:48.51 | [TK]D-Fender | Aziraphale, disable reinvives and for RTP through * |
| 19:49.58 | [TK]D-Fender | unbkbl, "a PBX" huh? Care to elaborate on that? |
| 19:51.16 | Aziraphale | [TK]D-Fender: cheers, will try that |
| 19:52.44 | dFence | *GARRR GARRR GARR* |
| 19:52.57 | dFence | ManxPower: i finally figured it out |
| 19:54.07 | dFence | behind a t-octopus you have to set the CallerID according to the assigned msn - didn't need that at home when it was hooked up to the pstn directly >_< |
| 19:54.25 | errr | [TK]D-Fender: when looking at: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+ExtensionState What is the ActionID its talking about? Is this the same as the status code? |
| 19:55.12 | [TK]D-Fender | errr, nope, but no idea what it really means. |
| 19:55.14 | unbkbl | the asterisk server is in another city... all the extensions attached to this server work fine, but i've configurated a new extension that can make calls but cannot recive calls, but i think now that the problem is because the phone is behind a router and it seems that is not allowing the incomming calls |
| 19:55.23 | errr | [TK]D-Fender: ok thanks |
| 19:55.41 | unbkbl | as deeperror said |
| 19:56.13 | [TK]D-Fender | unbkbl, read up : |
| 19:56.15 | [TK]D-Fender | ~sipnat |
| 19:56.16 | 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 |
| 19:56.17 | [TK]D-Fender | ^^^^^^^^^^^ |
| 19:56.35 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:57.11 | unbkbl | thnx so much for your attention i'll give it a look! |
| 19:58.49 | *** join/#asterisk Justnulling2 (n=Justnull@ool-457bcfaa.dyn.optonline.net) |
| 19:59.15 | _ShrikE | Is adtran rebranding polycom phones? |
| 19:59.48 | Qwell | dunno, but they wouldn't be the first. got an example? |
| 20:00.30 | _ShrikE | http://tinyurl.com/5lj7yj |
| 20:00.36 | Justnulling2 | keep getting rtc: lost some interrupts at 1024Hz., how can i fix it? |
| 20:00.51 | _ShrikE | The 700 series looks pretty sweet though |
| 20:00.51 | Qwell | _ShrikE: oh...well...clearly. |
| 20:00.52 | _ShrikE | http://www.adtran.com/adtranpx/Rooms/DisplayPages/layoutInitial?Container=com.webridge.entity.Entity%5bOID%5bE78C42554229154498899802662A989C |
| 20:01.47 | Qwell | _ShrikE: yeah, I don't know about the 700s, but clearly the rest are |
| 20:02.37 | unbkbl | another question, im trying compile zaptel in a new machine and i get the "You do nod appear to have the sources for the 2.6.18-6-486 kernel installed" error but when i do uname -r it shows 2.6.18-6-486, wich means it is alredy installed what can i do? |
| 20:02.54 | sp00kz | those are just rebranded polycoms other than the 706 |
| 20:02.55 | Qwell | unbkbl: uname -r doesn't show anything about kernel sources |
| 20:02.59 | [TK]D-Fender | _ShrikE, Clearly all the ones with Polycom model #'s are |
| 20:03.15 | _ShrikE | yeah |
| 20:03.41 | Aziraphale | unbkbl: you need to install kernel-devel |
| 20:03.53 | Aziraphale | you have the binary but not the sources |
| 20:04.01 | unbkbl | ok i'll try that |
| 20:04.59 | Justnulling2 | anyone has any ideas -> rtc: lost some interrupts at 1024Hz. |
| 20:06.20 | hardwire | actually, nobody has ideas on that one. |
| 20:06.30 | hardwire | I mean, lots of ideas - few conclusions. |
| 20:06.44 | [TK]D-Fender | _ShrikE, Cute models for their original stuff. Might be interesting to see if they're decent |
| 20:07.04 | [TK]D-Fender | Justnulling2, You need to set your timer for 1000hz, not 1024hz |
| 20:07.57 | Justnulling2 | [TK]D-Fender: how? where? |
| 20:08.04 | _ShrikE | [TK]D-Fender: I really do like adtran products as a whole. I think I am going to put my hands on one of those 700's |
| 20:08.40 | [TK]D-Fender | Justnulling2, kernel option. not sure where to set |
| 20:09.13 | Justnulling2 | [TK]D-Fender: hmm so need to recompile the kernel fo it? |
| 20:09.36 | [TK]D-Fender | Justnulling2, Don't know if you can tweak that another way or not... |
| 20:09.52 | [TK]D-Fender | Justnulling2, Google-able I'm sure. |
| 20:10.04 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 20:10.30 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 20:10.46 | *** join/#asterisk sp00kz (i=ilubj00@our.government.is.in.the.dark.bz) |
| 20:11.13 | *** join/#asterisk jmardonesk (n=jmardone@236-147-75.adsl.din.tie.cl) |
| 20:11.55 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 20:12.31 | Justnulling2 | [TK]D-Fender: google didn't help much and it works fine on my other machine with stock debian kernel so it is not it |
| 20:14.04 | jmardonesk | hello, Im looking to integrate mysql with asterisk to make surveys. I found DBqu from YOSD software, but I dont know another solution, Is this the best way for integrate mysql querys in the dialplan to save the survey answers. |
| 20:14.17 | rgsteele||work | Forgive me for the potentially uneducated question.... a typical T1 provides 23 PRI channels and 1 data channel. That's simultaneous calls, though, right? So, if your organization had 24 people, everyone would have to be on a call for the 24th person to run into issues? |
| 20:14.22 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:14.31 | Qwell | DBqu? never heard of it. What's it do? |
| 20:14.36 | Qwell | but - no. |
| 20:14.40 | jmardonesk | DBquery |
| 20:14.53 | jblack | rgsteele||work: usually correct. |
| 20:14.53 | Qwell | is that an Asterisk module, or something? |
| 20:15.19 | [TK]D-Fender | Justnulling2, looks like you need to recompile the kernel |
| 20:15.30 | jmardonesk | is an asterisk aplication, see http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DBQuery for more detalis |
| 20:15.37 | Qwell | jmardonesk: func_odbc |
| 20:15.38 | rgsteele||work | jblack: Thanks. I'm looking into phasing out the current POTS setup in favor of VOIP, trying to get a handle on everything before embarking down that path. |
| 20:15.53 | [TK]D-Fender | rgsteele||work, 24th call on your PRI. |
| 20:16.14 | lesouvage | Shouldn't a "make clean" not erase all .o files and modules. I got a message "Your Asterisk modules directory, located at |
| 20:16.16 | rgsteele||work | [TK]D-Fender: Hm? |
| 20:16.16 | jmardonesk | Qwell, OK, I see.. |
| 20:16.22 | [TK]D-Fender | rgsteele||work, Don't over-associate a call as having to be bridged to another device. |
| 20:16.44 | lesouvage | contains modules that were not installed by this |
| 20:16.46 | lesouvage | <PROTECTED> |
| 20:17.13 | rgsteele||work | [TK]D-Fender: Ok, I meant 23 calls in which two endpoints are connected and the parties are conversing. |
| 20:17.17 | [TK]D-Fender | rgsteele||work, and PRI != VoIP |
| 20:17.43 | [TK]D-Fender | rgsteele||work, * can be talking to 23 incoming calls from your PRI with NO "phones" in use at all. Its still just a "call" |
| 20:17.44 | dFence | what's the syntax for ImportVar when used with a call-file and setvar? |
| 20:17.53 | Aziraphale | rgsteele||work: can be connected to an asterisk box though |
| 20:17.58 | lesouvage | Sorry for passing my question in a strange way. |
| 20:18.32 | [TK]D-Fender | rgsteele||work, 23 PSTN channels. Doesn't matter if * wants to bridge them to anything. * itself is talking to that channel. |
| 20:18.57 | rgsteele||work | Ah, okay. So, if I've got PRI going over my T1, and > 23 people called in simultaneously, I'd run in to issues. |
| 20:19.13 | rgsteele||work | Even if they weren't connected to a warm body in the office? |
| 20:19.22 | *** join/#asterisk Zar0 (n=J0ff@modemcable119.221-56-74.mc.videotron.ca) |
| 20:19.23 | [TK]D-Fender | rgsteele||work, a call is a call is a call. |
| 20:19.42 | [TK]D-Fender | rgsteele||work, they're all just channels, so yeah, #24 gets a "busy" |
| 20:19.45 | Aziraphale | 23 ringing out and 24 gets a busy |
| 20:19.46 | rgsteele||work | [TK]D-Fender: Understood. |
| 20:20.11 | rgsteele||work | [TK]D-Fender, Aziraphale: Thanks for dispelling the fog. |
| 20:21.09 | rgsteele||work | I suppose then, if I thought my call volume would exceed that, I'd need PRI services running over either multiple T1's or a T2/T3? |
| 20:21.34 | [TK]D-Fender | rgsteele||work, thats definitely a way |
| 20:21.34 | Zar0 | Lets say I have 2 normal POTS lines at home. Is it possible to use my asterisk box with I guess a dual FXO card to take an incoming call from one of the normal POTS line and then use the other POTS line to dial out and connect that incoming call? |
| 20:21.54 | rgsteele||work | Or some other connection with greater bandwidth. [TK]D-Fender: There is another way? |
| 20:21.54 | [TK]D-Fender | Zar0, Sure |
| 20:22.15 | Zar0 | TKD-Fender: What kind of card do I need? A dual FXO card? |
| 20:22.16 | [TK]D-Fender | rgsteele||work, More T1's, VoIP, etc. |
| 20:22.29 | [TK]D-Fender | Zar0, any way to get 2 FXO's to * |
| 20:22.43 | Aziraphale | Zar0: done something like that with VoIP + GSM adapter :) |
| 20:22.51 | rgsteele||work | [TK]D-Fender: Hm, apparently your hint above didn't sink in until just now, that PRI != VoIP. |
| 20:22.57 | hardwire | I'd love a GSM adapter |
| 20:22.59 | hardwire | and service |
| 20:23.02 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 20:23.04 | hardwire | for free plz |
| 20:23.16 | [TK]D-Fender | rgsteele||work, PRI is a signalling over T1 which is TDM, not VoIP |
| 20:23.21 | Zar0 | TKD: SO the card Digium TDM402E 2FXO Analog TDM PCI Card would be fine to achieve what I want? |
| 20:23.35 | [TK]D-Fender | Zar0, that'd do. |
| 20:23.44 | Zar0 | Thanks |
| 20:23.46 | rgsteele||work | [TK]D-Fender: Ah, which would require a channelbank or a card in the * box to convert the lines from analog to digital? |
| 20:24.01 | [TK]D-Fender | rgsteele||work, T1 *is* digital.... |
| 20:24.23 | [TK]D-Fender | rgsteele||work, and you'd buy a T1 interface of some kind for * |
| 20:24.37 | rgsteele||work | Er, yeah I'm sorry that was a mistake. I meant some kind of card to convert the T1 into something the * box can deal with. |
| 20:24.58 | [TK]D-Fender | rgsteele||work, Digium and Sangoma are the 2 biggest and most use makers of PC cards for this purpose |
| 20:25.11 | rgsteele||work | [TK]D-Fender: I think I want to go the VoIP route. |
| 20:25.30 | [TK]D-Fender | rgsteele||work, Zaptel is the package that lets * use telephony cards. |
| 20:25.54 | [TK]D-Fender | rgsteele||work, there are ups & downs with VoIP. what are your actual needs? |
| 20:26.24 | rgsteele||work | [TK]D-Fender: I have an office of ~30 individuals. I just need something more flexible than the current Bizfon units provide. |
| 20:27.07 | rgsteele||work | I need to be able to customize the order in which the phones ring, provide conferencing, etc. I'm not designing a call center, I just need to fit the needs of a growing small business. |
| 20:27.48 | [TK]D-Fender | rgsteele||work, and what kind of PSTN connectivity do you have currently? |
| 20:28.43 | [TK]D-Fender | looks like Bizfon is rebranding Snom phones. What is this, National Rebranding Day? |
| 20:29.06 | rgsteele||work | We've got 6 lines. |
| 20:29.34 | rgsteele||work | Basic features on them, no big thrills or frills. |
| 20:29.48 | jaytee | we should have a National 2 for 1 Day where everything you buy on that day is half-off. |
| 20:29.56 | [TK]D-Fender | rgsteele||work, So why are you looking BEYOND a 23-port solution already? |
| 20:31.25 | jblack | jaytee: Boo. |
| 20:31.39 | jblack | They'll just double the prices every other day of the year. |
| 20:32.11 | rgsteele||work | [TK]D-Fender: I just want to be prepared for growth. I guess I'm not sure whether or not I should start right off the bat with VoIP, or start with PRI over T1, and have to deal with the bandwidth issues in the future if and when we outgrow the 23 channels. |
| 20:33.07 | [TK]D-Fender | rgsteele||work, You thinking you'll need to support over 4x as man calls as you do now? |
| 20:33.29 | [TK]D-Fender | jblack, 50% off 200% of the original price... a STEAL! |
| 20:33.36 | rgsteele||work | [TK]D-Fender: Depends on how top-heavy the grand poobahs make this place :) |
| 20:33.41 | jblack | yup |
| 20:34.26 | [TK]D-Fender | rgsteele||work, VoIP might cost less in some cases, more in others, and is definitely more trouble. |
| 20:35.00 | [TK]D-Fender | rgsteele||work, price out a full & partial PRI where you are and compare VS what it'd take to go VoIP, and support it. |
| 20:36.32 | rgsteele||work | [TK]D-Fender: I like that VOIP (seems like it) scales better, but more trouble isn't what I'm looking for. What types of issues are prevalent with VoIP that aren't as much of a problem with the PRI solution? |
| 20:37.08 | *** join/#asterisk robevans (n=robevans@OL6-231.fibertel.com.ar) |
| 20:37.51 | [TK]D-Fender | rgsteele||work, Internet failures of every kind. bandwidth. Security. Jitter. Packet-loss. Cost of paying for your internet connection itself AND VoIP service. |
| 20:38.11 | [TK]D-Fender | rgsteele||work, And that assuming your ITSP will have good audio quality, etc all of their own. |
| 20:40.21 | rgsteele||work | Hm. Don't bandwidth, packet loss, and jitter still occur with the PRI over T1 solution, too? |
| 20:40.30 | rgsteele||work | I don't see how that's specific to VoIP. |
| 20:41.03 | jblack | Dedicated lines makes all the difference |
| 20:41.14 | jblack | There's no contention for bandwidth on a PRI |
| 20:41.29 | [TK]D-Fender | rgsteele||work, T1 is a direct clocked link to the telco, not *internet* |
| 20:41.40 | *** part/#asterisk kerill (n=blades20@85.134.139.148) |
| 20:41.52 | [TK]D-Fender | rgsteele||work, T1 is TDM, not Packet-based |
| 20:42.02 | rgsteele||work | Ah... that makes more sense. |
| 20:42.31 | [TK]D-Fender | rgsteele||work, Packets get lost, mangled & misplaced. Time is constant. |
| 20:43.15 | file | lost packets are sad packets |
| 20:44.29 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 20:45.37 | rgsteele||work | [TK]D-Fender: Hm, okay that makes more sense. I certainly value reliability in this equation higher than anything else. Sounds like PRI T1 is a viable solution... I'd just need to narrow down which T1 card is right. Maybe I'll take a look at the Digium's out there - I have heard good things about them. |
| 20:46.06 | rgsteele||work | jblack, [TK]D-Fender: I do appreciate the advice. |
| 20:46.32 | *** join/#asterisk b11d` (n=no@234-200-29-134.hcc.mnscu.edu) |
| 20:47.10 | b11d` | what so Zaptel is getting renamed to DAHDI?? |
| 20:47.23 | *** join/#asterisk exothermc (n=miles@74.85.89.146) |
| 20:47.37 | [TK]D-Fender | rgsteele||work, I would suggest a 2 or 4 port card with hardware echo-cancelation and in PCI for maximum interoperability |
| 20:47.40 | *** join/#asterisk ccvp (n=ccvp@66.0.46.210) |
| 20:47.51 | ccvp | hello fellow internet addicts |
| 20:47.56 | [TK]D-Fender | b11d`, load pbx_whosyourDAHDI.so ;) |
| 20:47.58 | ccvp | how was your day of internet / irc addiction? :) |
| 20:48.01 | b11d` | DOH |
| 20:48.03 | b11d` | jhahahaha |
| 20:48.07 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 20:48.22 | rgsteele||work | [TK]D-Fender: Any specific model recommendations? |
| 20:48.23 | ccvp | fender, i heard ThakillaZ |
| 20:48.26 | ccvp | wiped alot out today :) |
| 20:49.43 | *** join/#asterisk CaTtleyA (n=DDoS@bas1-montrealak-1128581558.dsl.bell.ca) |
| 20:50.28 | [TK]D-Fender | rgsteele||work, I have a personal preference for the Sangoma A10(2/4)d |
| 20:50.46 | exothermc | What do I need to do to get asterisk to pass h.263? |
| 20:50.48 | [TK]D-Fender | rgsteele||work, www.telephonydepot.com |
| 20:51.03 | b11d` | so.. honestly though.. Zaptel will be called DAHDI? |
| 20:51.03 | [TK]D-Fender | exothermc, lookup "asterisk video" on the WIKI. |
| 20:51.16 | [TK]D-Fender | b11d`, Where do you see this? |
| 20:51.20 | b11d` | on asterisk.org |
| 20:51.27 | b11d` | in the News section |
| 20:51.52 | exothermc | I have videosupport=yes and allow=h263 and allow=h263p. I can see the h263 going to the asterisk IP but that is all I see. |
| 20:52.13 | *** join/#asterisk mindCrime (n=chatzill@216.27.62.2) |
| 20:52.36 | [TK]D-Fender | exothermc, What else are you expecting? |
| 20:52.39 | robevans | Anyone know of a service that will take my phone number list and make it available for sale but keep the number anonymous? |
| 20:52.55 | [TK]D-Fender | exothermc, pastebin the CLI & sip debug of a complete call, end-to-end along with your sip.conf |
| 20:53.00 | exothermc | [TK]D-Fender: For it to pass the h263 to the other end point which it doesn't |
| 20:53.00 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-10-21-ndn-esr-2.dynamic.isadsl.co.za) |
| 20:53.17 | exothermc | [TK]D-Fender: Right now asterisk is a blackhole for h263 |
| 20:53.30 | [TK]D-Fender | exothermc, provide the info I jsut requested. |
| 20:53.38 | exothermc | [TK]D-Fender: putting it together now. |
| 20:53.45 | exothermc | [TK]D-Fender: debug level 10 fine? |
| 20:54.12 | [TK]D-Fender | exothermc, verbose 10, sip debug, core debug 0 |
| 20:55.40 | *** join/#asterisk uluatu (n=deg@200.195.161.164) |
| 20:57.10 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:57.20 | exothermc | debug 0? as in off? |
| 20:58.13 | [TK]D-Fender | exothermc, Correct |
| 20:58.27 | exothermc | http://www.pastebin.ca/1023170 |
| 20:58.39 | exothermc | I had it on, but it isn't that messy. |
| 20:58.53 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 20:59.09 | exothermc | both end points were set to use h.263 |
| 20:59.19 | [TK]D-Fender | exothermc, Now try to provide everything I actaully asked for... |
| 20:59.34 | exothermc | packet captures on the asterisk node shows h263 coming in from both end points. |
| 20:59.41 | exothermc | [TK]D-Fender: rereading it. |
| 21:00.26 | exothermc | [TK]D-Fender: Could you clue me in on what piece is missing? |
| 21:01.08 | exothermc | oh I see it now |
| 21:01.09 | [TK]D-Fender | exothermc, sip debug, and your exact sip.conf masking only passwords for all ends involved. |
| 21:01.15 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 21:01.49 | exothermc | [TK]D-Fender: It is possible for peers not involved in the call to effect peers that are? |
| 21:02.59 | [TK]D-Fender | exothermc, No. [general] + peers please... |
| 21:04.05 | exothermc | [TK]D-Fender: never used the sip debug before, is there a nice way to pipe that to a file, and limit the hosts that it is watching? |
| 21:04.26 | [TK]D-Fender | exothermc, "sip debug peer [peername]" |
| 21:04.34 | [TK]D-Fender | exothermc, No [] in there. |
| 21:04.49 | exothermc | ya, and for multiple peers? |
| 21:05.34 | exothermc | [TK]D-Fender: or are you just interested in one of the end points at a time? |
| 21:05.39 | [TK]D-Fender | exothermc, multiple calls to that |
| 21:05.58 | [TK]D-Fender | exothermc, I want both ends of this single call. |
| 21:06.28 | exothermc | Ok but how do I limit debug to two peers? |
| 21:08.38 | TrentCreek | n |
| 21:09.01 | [TK]D-Fender | exothermc, only ENABLE it for 2 peers |
| 21:09.13 | exothermc | ahh ok got it |
| 21:09.31 | exothermc | I thought that was the filter not that it was on/off for each peer |
| 21:10.26 | Katty | yawns |
| 21:14.17 | *** join/#asterisk VaNNi (n=VaNNi___@lgb-static-216.70.165.200.mpowercom.net) |
| 21:16.50 | *** join/#asterisk wonderworld (n=voici@ip-62-143-31-149.hsi.ish.de) |
| 21:16.50 | _ShrikE | catches Katty's contagious yawns. |
| 21:17.24 | wonderworld | hi, i am trying to execute a command from an agi script. this is what i get on the asterisk-console: |
| 21:17.25 | wonderworld | -- AGI Script Executing Application: (MP3Player(http://some.live.stream/stream.mp3)) Options: ((null)) |
| 21:17.25 | wonderworld | May 19 23:15:35 WARNING[19250]: res_agi.c:1101 handle_exec: Could not find application (MP3Player(http://some.live.stream/stream.mp3)) |
| 21:17.45 | wonderworld | show applications lists MP3Player.... |
| 21:17.51 | wonderworld | what might be wrong? tnx. |
| 21:18.25 | exothermc | [TK]D-Fender: http://www.pastebin.ca/1023189 |
| 21:18.37 | exothermc | [TK]D-Fender: Does that work better for you? |
| 21:19.54 | rgsteele||work | [TK]D-Fender: If I understand it correctly, the only difference between the A101D and the A102D is I'd be able to handle only 30 calls versus 60? |
| 21:20.22 | [TK]D-Fender | rgsteele||work, 2 ports instead of 1, correct. |
| 21:20.39 | [TK]D-Fender | rgsteele||work, and 30/60 implies E1 PRI, not T1 PRI |
| 21:20.42 | [TK]D-Fender | ~e1 |
| 21:20.44 | jbot | [~E1] E1 is the basic digital telephony circuit used everywhere except the US,Japan,Taiwan & Hong Kong (who use T1 and similar variants). E1 runs at 2.048Mbps. It can be an unstructured channel for data. It can be channelized, to provide 31 time slots of voice or data, each of 64kbps. Time slot 16 is used for D-Chan when used with PRI signalling. |
| 21:21.22 | rgsteele||work | [TK]D-Fender: Ah, so 23/46 is correct, then? |
| 21:21.44 | [TK]D-Fender | rgsteele||work, yes |
| 21:21.55 | rgsteele||work | Good to see basic math skills are available to me past 5pm :P |
| 21:23.12 | [TK]D-Fender | Only 3 kinds of people in this world ; those that know math, and those that don't. |
| 21:23.44 | exothermc | [TK]D-Fender: Any idea from that paste why asterisk isn't passing the h263? |
| 21:25.03 | [TK]D-Fender | exothermc, not goo enough. I said I want the entire call from beginning to end. |
| 21:25.24 | exothermc | [TK]D-Fender: hmm I thought was what I grabbed. |
| 21:25.36 | [TK]D-Fender | exothermc, very clearly wrong. |
| 21:26.25 | [TK]D-Fender | exothermc, Capabilities: us - 0x18000c (ulaw|alaw|h263|h263p), peer - audio=0x8000c (ulaw|alaw|h263)/video=0x80000 (h263), combined - 0x8000c (ulaw|alaw|h263) |
| 21:26.45 | [TK]D-Fender | exothermc, but it does seem to agree on H.263 on both sides. |
| 21:26.50 | [TK]D-Fender | exothermc, check your clients. |
| 21:28.24 | exothermc | [TK]D-Fender: Checking now, here is the new paste http://www.pastebin.ca/1023201 |
| 21:29.01 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 21:29.01 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 21:30.53 | exothermc | [TK]D-Fender: I do see what you are saying though |
| 21:31.37 | [TK]D-Fender | exothermc, make sure your clients are set to initiate video on the call. Try toggling it during it. |
| 21:32.20 | exothermc | [TK]D-Fender: This is eyebeam, I start the call, then turn on the video feed. |
| 21:38.46 | exothermc | [TK]D-Fender: So unless h263 is negotiated then asterisk will ignore it? |
| 21:39.35 | [TK]D-Fender | exothermc, should. And it does seem to accept on both side. |
| 21:41.15 | exothermc | both sides? client to B2BUA or client to client? |
| 21:41.34 | exothermc | or either side of the B2BUA |
| 21:44.18 | *** join/#asterisk Justnulling2 (n=Justnull@ool-457bcfaa.dyn.optonline.net) |
| 21:46.21 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 21:46.21 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 21:50.32 | [TK]D-Fender | exothermc, on each side on *. |
| 21:50.34 | Justnulling2 | recompiled zaptel now i get in addition this error -> rc_avpair_new: unknown attribute 1490026597 |
| 21:50.39 | [TK]D-Fender | exothermc, so it does seem fine. |
| 21:54.03 | Justnulling2 | [TK]D-Fender: any idea about -> rc_avpair_new: unknown attribute 1490026597 |
| 21:55.58 | [TK]D-Fender | Justnulling2, pastebin is your friend.... |
| 21:59.56 | [TK]D-Fender | Justnulling2, when in doubt JFGI |
| 22:02.09 | Qwell | avpair is a freeradius thing |
| 22:03.15 | Justnulling2 | [TK]D-Fender: http://pastebin.ca/1023229 |
| 22:04.21 | Qwell | Justnulling2: asterisk is your hostname, isn't it? |
| 22:04.46 | Qwell | nm, misread. |
| 22:04.58 | Qwell | sounds like your freeradius/unixodbc configs are wrong though |
| 22:05.29 | Justnulling2 | [TK]D-Fender: google comes with 8 pages, 1 doesn't load 2 are icr logs, then there is spanish german and russian sites so no luck there |
| 22:06.24 | Justnulling2 | Qwell: it got this error after i recompiled zapata, how do i fix freeradius/unuxodbc config? |
| 22:06.37 | Qwell | it's not a zaptel problem |
| 22:07.20 | [TK]D-Fender | Justnulling2, Well thats a warning. Whats the PROBLEM? |
| 22:08.57 | Justnulling2 | [TK]D-Fender: zaptel is the problem mostly, tired installing the latest version and still get the rtc error and now on top of this i get rc_avpair_new: unknown attribute 1490026597 |
| 22:09.27 | ManxPower | Justnulling2: RTP error? |
| 22:09.31 | Justnulling2 | qwell: well this happend after i installed latest zaptel, it is still neded with 2.6? |
| 22:09.34 | [TK]D-Fender | Justnulling2, and I told you to fix your clock from 1024 to 1000 |
| 22:09.46 | [TK]D-Fender | Justnulling2, Who said that message had anything to do with anything? |
| 22:12.17 | ManxPower | Justnulling2: STOP saying "latest zaptel" Everything has a version, use it when talking about it. |
| 22:12.33 | ManxPower | Especially since there are at least three totally different zaptels that could be called "latest" |
| 22:13.09 | Justnulling2 | [TK]D-Fender: you did but googling it doesn't come up with anything that says need to recompile kernel for zapata to work, so don't want to touch my kernel |
| 22:14.01 | Justnulling2 | ManxPower: zaptel-1.4.10.1 didn't see there where other versions on the website, is this one i should be using? |
| 22:14.06 | ManxPower | Justnulling2: Asterisk doesn't care what you want or don't want. If [TK]D-Fender says you must recompile you kernel to fix the problem then than is what you need to do to fix the problem. |
| 22:14.13 | [TK]D-Fender | Justnulling2, let me reiterate this, one last time : If your RTC clock is set for andything other than 1000hz ZTDUMMY is *fucked*. Is that clear? |
| 22:14.26 | ManxPower | Justnulling2: look at the top of your screen in the area for the channel topic. |
| 22:16.06 | ManxPower | [TK]D-Fender: I keep telling you, if they don't listen, put the on /ignore |
| 22:16.15 | ManxPower | them too |
| 22:16.17 | [TK]D-Fender | Justnulling2, and feel free to "not want to touch my kernel". You can also "not want to cross the finish line". Just don't expect to win the race. |
| 22:16.57 | ManxPower | Eventually they will either start to follow the advice given them or they will be on everyone's /ignore list. Either way we win. |
| 22:17.08 | *** join/#asterisk Georger86 (n=ge@88.218.30.85) |
| 22:17.13 | Georger86 | hello |
| 22:17.18 | [TK]D-Fender | ManxPower, I have fortunately never had to put anyone on /ignore . |
| 22:17.31 | Georger86 | i need some asterisk assistance |
| 22:17.48 | ManxPower | [TK]D-Fender: I don't HAVE to either, but it's much less stressful if you do. |
| 22:17.50 | [TK]D-Fender | ~ask |
| 22:17.50 | jbot | somebody said ask was Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 22:18.12 | [TK]D-Fender | ManxPower, it would if I didn't care to help people period. |
| 22:18.48 | ManxPower | When we lose our temper it makes the whole channel look bad. No matter how much of a moron the other person is. |
| 22:19.31 | ManxPower | Georger86: speak on the channel or don't speak at all |
| 22:19.46 | ManxPower | Georger86 just private /msg'd me. |
| 22:20.10 | Georger86 | ok |
| 22:20.15 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 22:20.16 | Georger86 | im first time there |
| 22:20.17 | ManxPower | [TK]D-Fender: I try to stop caring about helping a person about the same time as they stop listening to me. |
| 22:20.33 | Georger86 | well check this and tell me ur oppinion |
| 22:20.34 | Georger86 | Connected to Asterisk 1.4.19.1-BRIstuffed-0.4.0-RC1 currently running on elastix (pid = 2551) |
| 22:20.34 | Georger86 | Verbosity was 3 and is now 6 |
| 22:20.34 | Georger86 | <PROTECTED> |
| 22:20.34 | Georger86 | <PROTECTED> |
| 22:20.34 | Georger86 | <PROTECTED> |
| 22:20.44 | [TK]D-Fender | Georger86, and NEVER spam like that in here again |
| 22:20.45 | ManxPower | Georger86: stop. Look at the following text |
| 22:20.47 | ManxPower | ~pb |
| 22:20.47 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines 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 |
| 22:20.52 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 22:21.03 | Justnulling2 | [TK]D-Fender: what is this ztdummy is sued for? do i need to run it (be in this race as you put it) |
| 22:21.25 | [TK]D-Fender | Justnulling2, what are you using Zaptel for? |
| 22:21.37 | ManxPower | Georger86: the call is coming into the extensions.conf context [from-sgm], there is no extension line in that context to match 6982926300. |
| 22:22.00 | Justnulling2 | [TK]D-Fender: nothing don't have any digium hardware |
| 22:22.21 | *** join/#asterisk phix (n=threat@123-243-44-131.tpgi.com.au) |
| 22:22.21 | [TK]D-Fender | Justnulling2, if you're not using it, why are you installing it? |
| 22:22.24 | *** join/#asterisk l3jj (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 22:22.25 | Georger86 | so i should edit extensions.conf? |
| 22:22.45 | ManxPower | Georger86: yes. I think you might want to stop and take some time to read the Good Book |
| 22:22.47 | ManxPower | ~book |
| 22:22.48 | 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 --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 22:23.01 | [TK]D-Fender | Georger86, Your zapata.conf is sending the calls to "s" in a context that doesn't match with extensions.conf. |
| 22:23.12 | ManxPower | [TK]D-Fender: read more carefully |
| 22:23.15 | Justnulling2 | [TK]D-Fender: comes as dependent to pluto-asterisk package |
| 22:23.19 | Georger86 | ok thank you:) |
| 22:23.26 | [TK]D-Fender | ManxPower, oops |
| 22:23.36 | ManxPower | [TK]D-Fender: I misread often enough 8-) |
| 22:23.42 | [TK]D-Fender | Georger86, indeed, to an exten that doesn't match in some context, and then failing back a few times. |
| 22:24.07 | ManxPower | [TK]D-Fender: I wish it did not do the failover stuff |
| 22:24.09 | [TK]D-Fender | ManxPower, funny thing is I did process it right the first time and by the time I got to typing, oops, there it went :) |
| 22:24.25 | [TK]D-Fender | ManxPower, ditto. "default" = BS to support shmucks. |
| 22:24.39 | ManxPower | I guess "s" is hyperlinked in your head to that answer. |
| 22:24.51 | [TK]D-Fender | ManxPower, perhaps something like that. |
| 22:24.53 | *** join/#asterisk CaTtleyA (n=DDoS@bas1-montrealak-1128581558.dsl.bell.ca) |
| 22:25.18 | ManxPower | Like the first thing I do when setting up Asterisk is to make sure that the default context is INVALID |
| 22:25.41 | *** part/#asterisk Justnulling2 (n=Justnull@ool-457bcfaa.dyn.optonline.net) |
| 22:26.21 | [TK]D-Fender | ManxPower, looks like our friend there simply has no idea why he was even doing what he was doing. |
| 22:31.51 | maqr | [TK]D-Fender: i installed the sample configs, and none of them say "ztdummy" anywhere in them... so i'm not sure if asterisk is actually using it or not |
| 22:32.13 | *** join/#asterisk klimonso (n=eddy@91.73.203.98) |
| 22:32.39 | [TK]D-Fender | maqr, and they won't either. |
| 22:33.07 | [TK]D-Fender | maqr, do YOU know why * would be using it in your setup? |
| 22:33.23 | klimonso | when i call in on inboud to asterisk and i am trying to dial an extention it doesnt even let me push it all, it takes me directly to the menu selected by ivr. where i can change the time? in freepbx??? |
| 22:33.38 | [TK]D-Fender | klimonso, ... |
| 22:33.40 | [TK]D-Fender | ~freepbx |
| 22:33.41 | jbot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 22:34.16 | klimonso | thx guys |
| 22:34.52 | maqr | [TK]D-Fender: something about timings or music-on-hold or something? so.... no, i have no idea, but i was under the impression that if you don't have ztdummy or some other timing hardware, something won't work as well as it could, or something |
| 22:36.46 | *** join/#asterisk sp00k3y (n=chatzill@74.202.4.2) |
| 22:38.39 | *** join/#asterisk terracon (n=greisky@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com) |
| 22:38.48 | [TK]D-Fender | maqr, go find out WHY you think you need it. Then when you do feel free to ask again later. |
| 22:39.01 | maqr | [TK]D-Fender: okay :) |
| 22:39.42 | exothermc | [TK]D-Fender: Ok Here is a better sip trace http://pastebin.com/d330aecb5 It looks like the clients are negotiating correctly. |
| 22:39.54 | [TK]D-Fender | maqr, indeed MoH can sync better with a Zaptel timing source, but its not "required". Its needed for MeetMe, and IAX2 Trunk Mode. Thats it. |
| 22:40.15 | [TK]D-Fender | exothermc, and I told you that * seems to be set fine, go work with the clients. |
| 22:42.34 | exothermc | [TK]D-Fender: I have been, on line 313 you see jon's client negotiating h263. Then on line 455 you see miles renegotiating h263, but asterisk never sends any h263 to either client even though the h263 is being sent from both clients and reaching asterisk. |
| 22:43.25 | exothermc | [TK]D-Fender: I'm watching a tcpdump on asterisk and I can see h263 coming in, but never leaving. |
| 22:46.15 | *** join/#asterisk Justnulling2 (n=Justnull@ool-457bcfaa.dyn.optonline.net) |
| 22:46.47 | *** join/#asterisk denon (n=denon@tooth.decay.org) |
| 22:46.47 | *** mode/#asterisk [+o denon] by ChanServ |
| 22:48.50 | [TK]D-Fender | exothermc, make sure the ports its using aren't being blocked. |
| 22:48.57 | [TK]D-Fender | exothermc, beyond that * seems fine |
| 22:49.07 | [TK]D-Fender | exothermc, so its wither networking ro your clients. |
| 22:49.10 | [TK]D-Fender | eiterh* |
| 22:49.12 | [TK]D-Fender | either* |
| 22:49.17 | [TK]D-Fender | bleh, can't type today... |
| 22:49.51 | exothermc | [TK]D-Fender: A you hearing that I'm saying there is no h263 leaving the asterisk box? (confirmed from tcpdump) Checking iptables now. |
| 22:50.44 | exothermc | [TK]D-Fender: iptables are empty |
| 22:51.07 | exothermc | so if asterisk was passing any h263 it should show up in tcpdump |
| 22:53.21 | [TK]D-Fender | exothermc, Make sure you've done "canreinvite=no" for your peers, and thats the end of what I can suggest. |
| 22:53.33 | exothermc | [TK]D-Fender: ok thanks |
| 22:54.05 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 22:54.59 | exothermc | [TK]D-Fender: It was set in general, but I'll add it to the peers |
| 22:55.33 | *** join/#asterisk Justnulling2 (n=Justnull@ool-457bcfaa.dyn.optonline.net) |
| 22:56.10 | exothermc | [TK]D-Fender: Should it be opening up a separate set of channels for the h263? |
| 22:56.35 | [TK]D-Fender | exothermc, 1 port per side for audio, another for video. |
| 22:57.16 | exothermc | [TK]D-Fender: hmm ya no video channels are opening up. |
| 22:57.49 | exothermc | [TK]D-Fender: What clients have you seen this working with yourself? |
| 22:58.37 | [TK]D-Fender | exothermc, I've used eyebeam myself. |
| 23:01.17 | maqr | is there some documentation for the sample configs that i'm unaware of? the book doesn't seem to cover them, and i'm having trouble understanding them |
| 23:11.37 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 23:14.01 | [TK]D-Fender | maqr, ... |
| 23:14.03 | [TK]D-Fender | ~book |
| 23:14.03 | 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 --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 23:14.05 | [TK]D-Fender | ~wikis |
| 23:14.05 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 23:19.51 | *** join/#asterisk Justnulling2 (n=Justnull@ool-457bcfaa.dyn.optonline.net) |
| 23:24.14 | *** join/#asterisk RoyK (n=roy@ip-170-58-149-91.dialup.ice.no) |
| 23:24.42 | *** part/#asterisk RoyK (n=roy@ip-170-58-149-91.dialup.ice.no) |
| 23:27.01 | *** join/#asterisk TrentCreek (n=trent@cpe-70-116-111-122.rgv.res.rr.com) |
| 23:38.17 | *** join/#asterisk trnzmeta (n=bleh@iinet.guard.com.au) |
| 23:41.15 | *** join/#asterisk uluatu (n=deg@189.58.3.155.adsl.gvt.net.br) |
| 23:42.09 | Yourname` | Hi, Is there a way I can change the format of the email that Asterisk sends out when a voicemail is left? |
| 23:48.35 | _ShrikE | Cant you modify that in voicemail.conf? |
| 23:50.31 | Qwell | yes |
| 23:51.07 | [TK]D-Fender | load chan_zomgyoumeanthesampleconfigsprettymuchspeelitoutingoreydetail.so |
| 23:53.38 | Yourname` | _ShrikE: Found it, thanks.. |
| 23:56.18 | mwalling | [TK]D-Fender: heh |