| 00:24.53 | *** join/#asterisk b14ck (~rdegges@cpe-24-24-128-47.socal.res.rr.com) |
| 00:25.22 | Goshen | Does anyone know how to log into the Sipura 2102 admin pannel? just going to its DHCP assigned ip doesn't work, and I can't find the user manual online, the devices is not locked |
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| 00:45.12 | Beltechs | hello Im using *1.6 and Im having remote extension issues where the phones are unregistering. I ran sip set debug peer 6960. I am able to decipher I have posted on http://pastebin.ca/1908337 Thank you. |
| 00:45.24 | *** part/#asterisk iamy_china (~iamy_chin@221.221.149.166) |
| 00:45.43 | Beltechs | Fender u there? |
| 00:45.50 | Beltechs | u ther |
| 00:52.41 | Beltechs | This pasepin is of another extension that is constantly unregistering. http://pastebin.ca/1908345 |
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| 01:13.33 | Goshen | I remembered, to configure a Linksys spa2102 you have to plug your computer into the ethernet port |
| 01:21.27 | ManxPower | Or you can dial *** (IIRC) and get into the Configuration Menu |
| 01:22.59 | Goshen | **** |
| 01:25.03 | Goshen | Now I need to make a dialplan for the Sipura so it sends the dial to asterisk faster without waiting |
| 01:25.46 | Goshen | It waits 10 seconds before it sends the dial to Asterisk which is really annoying |
| 01:26.42 | Goshen | I am thinking that is because of the dialplan in the firmware |
| 01:27.54 | Kevin` | yep |
| 01:28.37 | Goshen | happen to know a dialplan so I don't have to figure it out? :) |
| 01:29.31 | ChannelZ | depends on what all you want to do |
| 01:30.00 | ChannelZ | ([2-9]xx[2-9]xxxxxx|911) would let you make 10-digit calls plus 911 in the US |
| 01:30.37 | ChannelZ | (asterisk extensions programmed accordingly) |
| 01:31.13 | Kevin` | mine is (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.|1xx|5xx|9|8) |
| 01:31.18 | Kevin` | but I forget what it does |
| 01:31.22 | Goshen | the asterisk extensions will be 102 or 103 |
| 01:31.37 | ChannelZ | so add 1xx |
| 01:31.39 | Kevin` | probably just matching stuff so it doesn't delay. everything goes through asterisk |
| 01:32.05 | Goshen | cool thanks! |
| 01:32.32 | Goshen | why did you add 9 and 8 to yours? |
| 01:32.41 | Kevin` | because i have a 9 and 8 extension |
| 01:32.47 | Goshen | ahh |
| 01:32.50 | Kevin` | which gives a dialtone on different lines |
| 01:32.55 | Goshen | aha |
| 01:33.26 | Goshen | thats a good idea |
| 01:33.36 | Goshen | so you can choose what outbound line you are going out on |
| 01:34.53 | drmessano | Choosing outbound lines or trunks by prefix is kinda silly |
| 01:35.57 | Kevin` | the purpose is to bypass asterisk's normal dialplan and enter a number directly on the line |
| 01:36.14 | Kevin` | I mostly use it for testing |
| 01:36.15 | drmessano | Dialplans should be built in Asterisk that make proper use of the distinction between different routes |
| 01:36.41 | Kevin` | how do YOU let a user choose a route for a call |
| 01:36.52 | drmessano | You don't |
| 01:37.08 | Kevin` | now, see, that's annoying sometimes :) |
| 01:37.19 | Goshen | I can see the logic in that though, because say you wanted the caller ID of the call to be a specific number, you need to do the prefix meathod |
| 01:37.31 | Kevin` | actually |
| 01:38.02 | Kevin` | if I want the caller id to be a specific number, I have an extension in asterisk that asks for a phone number, sets the caller id, then goes back to asking for a number to call |
| 01:38.07 | drmessano | If you have a need for a different outbound route, the dialplan should reflect that difference |
| 01:38.47 | Kevin` | again, mostly for testing |
| 01:38.54 | drmessano | For example, if I have a different provider for local calls, my dialplan should use that provider/line based on the dialed number. |
| 01:39.01 | Kevin` | I don't really do prank calls ;p |
| 01:39.17 | drmessano | heh |
| 01:39.31 | Kevin` | drmessano: of course. but what if you want to dial a local number through the other provider for some reason? |
| 01:39.44 | Kevin` | or the other way around |
| 01:41.04 | drmessano | Kevin`: That argument isn't enough of a reason to (1) make users dial a prefix before each call and (2) allow them to have that power. |
| 01:41.32 | Kevin` | of course they don't HAVE to dial the prefix |
| 01:41.39 | Kevin` | as to power, yes, but this is for me |
| 01:41.56 | drmessano | I'm not arguing about your testing extension |
| 01:41.58 | Kevin` | i'd prefer just pressing a number to fiddling with making a call manually on the console |
| 01:42.02 | drmessano | So what's the issue? |
| 01:43.10 | *** join/#asterisk ChannelZ (~bobm@burner.com) |
| 01:45.25 | Goshen | that didn't speed it up, it still waits 10 seconds before passing the call to Asterisk |
| 01:48.43 | *** join/#asterisk [netman] (~netman@83.54.228.88) |
| 01:57.35 | drmessano | Goshen, if the dialed digits match the dialplan, there wont be a delay |
| 01:59.22 | Goshen | I am reading the cisco documentation - it says to use S0 at the end of the dialplan, but it doesn't work |
| 02:06.54 | Goshen | aha, I had one too many X's :) |
| 02:07.13 | Goshen | fixed now :) |
| 02:07.26 | Gershwin | anyone here ever used or configured a nortel 1535? |
| 02:08.06 | Gershwin | this one here: http://www2.nortel.com/go/product_content.jsp?segId=0&catId=-9789&parId=0&prod_id=59520&locale=en-us |
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| 03:16.39 | ChannelZ | looks like an Ikeaphone |
| 03:17.06 | Gershwin | really? |
| 03:17.13 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 03:17.21 | Gershwin | I thought it looked more like a Macysphone or a Targetphone |
| 03:17.26 | Gershwin | i would have never guessed Ikea |
| 03:17.39 | xheliox | I had heard Ikea was investing heavily in the telephony market.. |
| 03:18.03 | xheliox | Their hardware echo canncelation algorithm is going to blow the doors off the market. I can't wait. |
| 03:18.27 | xheliox | Could triple call capacity on lower end servers. |
| 03:18.47 | ChannelZ | And when the phone rings, it emits a meatball smell |
| 03:19.20 | xheliox | That's really a feature that Polycom should pick up on. |
| 03:19.42 | xheliox | Every time the phone rings, I cry out.. "damn that lack of meatball or cookie smell!" |
| 03:20.56 | xheliox | I prefer the server.0.smell="chococlate_chip_cookie", but to each their own. |
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| 03:26.00 | *** join/#asterisk Goshen (~Goshen@c-174-52-7-122.hsd1.ut.comcast.net) |
| 03:26.52 | Goshen | I am trying to provision my IAXy 2 and using the linux that Asterisknow installed, but it doesn't have SVN, what do I type into yum to install svn so I can compile the iaxy software? |
| 03:27.22 | xheliox | lol - Goshen.. |
| 03:27.34 | xheliox | Did you go out and buy every random SIP/IAX device on the planet? |
| 03:27.47 | Goshen | :) no, I had stuff left over from my old install |
| 03:28.21 | Goshen | Polycom is working, Sipura is working, now I am playing with my IAXy |
| 03:28.21 | xheliox | Could be my imagination, but it seems like you've mentioned every ATA/voip phone on the market in the last 48 hours. |
| 03:28.24 | Goshen | I like this Polycom 303, I think I will get 10 more |
| 03:28.36 | xheliox | It's been a long long time since I had my IAXy in operation anywhere.. |
| 03:29.12 | xheliox | But I suspect you'll have to use svn to checkout the iaxyprov tool. I don't think I've seen it anywhere in a long time. |
| 03:29.43 | xheliox | I know the asterisk-trunk hasn't had the iaxy firmware published for many moons, you have to get it outside of the asterisk source and it's a very very discontinued product. |
| 03:30.07 | xheliox | Don't get me wrong, I respect the desire to experiment, just made me chuckle. :) |
| 03:30.22 | xheliox | And there's nothing all that wrong with an IAXy, I used one for quite some time at home. |
| 03:31.52 | hmodes | it's also the only UA on earth that supports pulse dialing :) |
| 03:33.24 | Goshen | I just need to know what the package name is called on centos that will give me an svn client |
| 03:34.30 | hmodes | subversion.i386 : Modern Version Control System designed to replace CVS |
| 03:34.43 | hmodes | from epel, I think |
| 03:35.44 | hmodes | ah, nope, rpmforge |
| 03:35.45 | hmodes | subversion-1.6.12-0.1.el5.rf |
| 03:37.02 | hmodes | http://dag.wieers.com/rpm/packages/rpmforge-release/ |
| 03:37.22 | hmodes | rpmforge and epel make centos actually useful ;p |
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| 03:48.57 | mutineer612 | hi |
| 03:49.24 | xheliox | Hello. |
| 03:49.30 | mutineer612 | I have been testing Avaya 96xx phones with Asterisk 1.6.2 running the SIP firmware. The phones register with Asterisk just fine and most basic functions are working correctly. However when I setup the same configuration using Switchvox "free" edition I'm unable to get the phone registered. Has anyone else tried an Avaya phone? Or have suggestions as to why Asterisk 1.6.2 works but Switchvox v9525 does not work? |
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| 04:00.06 | Gershwin | dunno.. but i bet a quick call to digium during business hours would help |
| 04:05.45 | mutineer612 | ok |
| 04:10.58 | [TK]D-Fender | mutineer612: Nothing changes with the registration process. |
| 04:11.16 | [TK]D-Fender | mutineer612: So some basic configuration is simply wrong on your Swithvox setup |
| 04:12.47 | Gershwin | good point |
| 04:13.07 | Gershwin | doesn't switchvox usu. lag asterisk by a few digits? |
| 04:13.23 | Gershwin | or no... i don't use sv btw, so dunno |
| 04:14.22 | [TK]D-Fender | lag is irrelevant for this |
| 04:16.05 | mutineer612 | hmm... my sip.conf is very simple... the only thing that I can think of that is different is that my Asterisk setup was 4 digit, and Switchvox is a 3 digit dial plan. |
| 04:16.13 | Gershwin | ok... i think what you're sayign is that the avaya 96xx have auto-provisioned for some time now |
| 04:19.09 | hmodes | sip set debug peer avaya |
| 04:19.55 | [TK]D-Fender | Dialplan has no impact on registration |
| 04:19.57 | Goshen | hmodes, thanks :) |
| 04:20.02 | hmodes | 9 times out of 10 the authentication is failing for a stupid reason :) |
| 04:20.14 | hmodes | you'll see it in the sip 401/403 |
| 04:24.25 | Goshen | I get this when trying to make the iaxyprov utility- make: cc: Command not found |
| 04:24.25 | Goshen | make: *** [provision.o] Error 127 |
| 04:24.55 | mutineer612 | ok, I'll have to get a sniffer setup... as I cant run the debug cmd from Switchvox server. It has no cmd line access. |
| 04:28.51 | [TK]D-Fender | mutineer612: Go change that. |
| 04:31.42 | hmodes | yeah, not being able to debug is fairly fail |
| 04:31.51 | xheliox | Goshen: yum install gcc :P |
| 04:33.49 | Goshen | xheliox, thanks |
| 04:34.47 | drfreeze | What does Answer() really do. Is it just a demarcation of call progress? |
| 04:35.30 | drfreeze | Because Background can be run with option 'n', which means that audio is being played before the call is answered |
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| 04:37.54 | hmodes | in the sip-world answer is the difference between a 180 session progress and 183 early media, I think |
| 04:37.58 | [TK]D-Fender | drfreeze: Because maybe you want to make sure the call doesn't fall through, and not all channels support early media, etc |
| 04:38.27 | hmodes | also that |
| 04:39.31 | hmodes | it mostly matters for billing systems |
| 04:40.45 | hmodes | generally if you 'answer' a call, billing begins, otherwise you can ring-through and not get charged |
| 04:41.03 | hmodes | at least that's been my experience |
| 04:41.58 | [TK]D-Fender | If you try early media with most ITSPs I'm sure you'll find yourself billed for that time. |
| 04:42.21 | [TK]D-Fender | Which will lead to billing mismatches |
| 04:42.38 | drfreeze | If an incoming call is sent to a sip phone with dial and answer is never called explicitly, does answer get implicitly called when the call is picked up? |
| 04:42.42 | [TK]D-Fender | Only benifit is for your caller... |
| 04:42.57 | [TK]D-Fender | drfreeze: Yes |
| 04:43.08 | [TK]D-Fender | drfreeze: Allowing end-end progress |
| 04:43.23 | [TK]D-Fender | drfreeze: HOWEVER you runt he risk of split call-timeout issues |
| 04:43.57 | [TK]D-Fender | drfreeze: By answering you can ring longer than your caller may permit allowing you to ring the dest longer |
| 04:44.37 | drfreeze | interesting |
| 04:45.04 | drfreeze | what if, instead of answer, I run Progress |
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| 04:45.21 | [TK]D-Fender | drfNo dodge on this one. Answer or do not.f |
| 04:45.26 | drfreeze | I assume that is meaningless to the ITSP |
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| 05:03.48 | xheliox | I'm unreasonably bored. |
| 05:03.55 | xheliox | And rather jet lagged, ironically. |
| 05:05.10 | hmodes | answer, or do not, that's quite deep |
| 05:07.12 | xheliox | TK is quite the philosopher. |
| 05:07.37 | xheliox | To dial or not to dial, that is the question. |
| 05:16.23 | [TK]D-Fender | Do, or do not. There is no try. </yoda> |
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| 05:31.30 | hmodes | if only more sip implementations understood this |
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| 06:06.24 | hmodes | holy crap it's another upc user |
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| 06:11.40 | hmodes | I think I scared him off |
| 06:24.10 | shamelessn00b | '-' |
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| 06:28.24 | gavimobile | hi folks, for good quality phone calls I would need high upload speed, is that correct? |
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| 06:45.32 | Marquel | morning |
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| 07:28.25 | gavimobile | good morning |
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| 07:51.43 | Marquel | how do i make automon-feature work? since i've haven't had any feature.conf up until now i'm afraid it's not even read during asterisk startup. where can i look further? asterisk is 1.2 and i can't update due to driver issues. |
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| 08:52.19 | d43mOn | hi |
| 08:52.44 | d43mOn | i have a problem with backup and restore |
| 08:53.09 | d43mOn | i use asterisk with freepbx |
| 08:53.21 | d43mOn | i backup the databases and files config |
| 08:53.48 | tzafrir_laptop | d43mOn, #freepbx |
| 08:54.00 | d43mOn | ok thanks |
| 09:08.58 | redax | hi |
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| 09:52.58 | tzafrir_laptop | http://blog.steve.org.uk/sysadmin_im_considered_harmful.html - and now check the site it mentions |
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| 10:35.10 | ankur_6997 | hi every one i am new to astrisk and i am trying to implement a IVR using asterisk but i am not sure weather multiple caller can be handled by the IVR when i connect alocal PSTN line number to my astrix box using a FXO card |
| 10:35.22 | ankur_6997 | ? |
| 10:35.42 | ankur_6997 | please make it clear |
| 10:36.48 | tzafrir_laptop | This is #asterisk, not #aterix :-) . Also, what you write will be easier to parse if you apply some occasional punction marks |
| 10:37.36 | tzafrir_laptop | An FXO line only handles a single caller at a time, right |
| 10:37.42 | hmodes | also a single fxo line is always a single fxo line |
| 10:37.47 | hmodes | yes, that |
| 10:38.17 | ankur_6997 | so only one caller will be handled at one time ? |
| 10:38.35 | hmodes | if you only have one line going in to the box, yes |
| 10:39.02 | ankur_6997 | i have tested my ivrs with callcentric 's number it can handle multiple users |
| 10:39.45 | tzafrir_laptop | Asterisk has no problem supporting multiple calls, provided the channel supports it |
| 10:39.47 | hmodes | presumeably callcentric delivers their calls with voip and don't immediately limit you per-channel |
| 10:39.55 | tzafrir_laptop | FXO does not |
| 10:40.06 | hmodes | ^what he said |
| 10:40.13 | tzafrir_laptop | If you want more than one call at a time, get multiple FXO lines |
| 10:40.13 | ankur_6997 | so what is the solution |
| 10:40.37 | ankur_6997 | ok but i want to use only one number |
| 10:40.47 | tzafrir_laptop | (or E1, or VoIP trunks, or whatever) |
| 10:40.53 | ankur_6997 | say 12345678 |
| 10:41.00 | hmodes | have your DID delivered over sip |
| 10:41.06 | hmodes | see: voip.ms, etc. |
| 10:41.55 | ankur_6997 | i have no idea about DID |
| 10:42.16 | ankur_6997 | i want this service for Indian number |
| 10:42.23 | hmodes | direct-inward dial, a phone number, basically |
| 10:42.27 | tzafrir_laptop | The issue here is not really Asterisk It's how that call is delivered to Asterisk |
| 10:43.09 | hmodes | I think voip is marginally illegal in india |
| 10:43.18 | hmodes | which is very unfortunate |
| 10:43.43 | ankur_6997 | yes but what i have to do for that many ivrs system are used in india |
| 10:44.49 | hmodes | so long as nobody notices, you're probably good.. I have no idea how one gets an india-did tho' |
| 10:45.07 | ankur_6997 | so what i have to do to achive this(multiple user supporting IVR system) using pstn lines ? |
| 10:45.11 | hmodes | you're probably best off just getting a number in another country and routing your calls that way |
| 10:45.22 | ankur_6997 | any idea ? |
| 10:45.56 | ankur_6997 | can you give me some links to do the same as you have suggested ? |
| 10:46.05 | hmodes | buy an E1 or look in to sip trunking over ipsec :) |
| 10:46.51 | hmodes | from what I gather, your mileage will vary either way |
| 10:47.25 | hmodes | ip connectivity in india looks to be pretty crap |
| 10:48.26 | hmodes | tho' I did manage a callcenter there without getting yelled at regularly by my boss, so it seems nobody actually notices ~300ms of lag, or they don't complain loud enough, anyway |
| 10:48.45 | *** join/#asterisk Benwa (~Benwa@unaffiliated/benwa) |
| 10:49.18 | Trixboxer | ankur_6997, get a TATA E1 PRI line |
| 10:49.47 | Trixboxer | call them and they will give you a 100 DID E1 pri line with 30 channel capacity |
| 10:51.14 | Trixboxer | and ankur_6997 losts his ans :) |
| 10:54.48 | hmodes | it's best not to sell E1s to people anyway :) |
| 10:55.50 | hmodes | tdm is for suckers |
| 10:56.16 | *** join/#asterisk godfather_ (~godfather@79.Red-88-6-254.staticIP.rima-tde.net) |
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| 11:16.01 | Trixboxer | hmodes, no option in india, no one ready to give T1 |
| 11:16.13 | tzafrir_laptop | T1? E1? |
| 11:20.50 | *** join/#asterisk Benwa (~Benwa@unaffiliated/benwa) |
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| 11:34.31 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 11:38.19 | *** join/#asterisk telnettech (~telnettec@216.49.139.56) |
| 11:40.11 | *** join/#asterisk E-bola (~bola@188.120.76.228) |
| 11:40.43 | E-bola | I keep getting Username/auth name mismatch when i try to have my polycom soundstation ip 6000 register with my asterisk server (1.4) |
| 11:41.03 | E-bola | Asterisk just keeps telling me that: chan_sip.c:8507 check_auth: username mismatch, have <107>, digest has <> |
| 11:41.57 | E-bola | Anybody know what might be the problem? |
| 11:44.20 | hrhrhr | how does a ring group actually work? is it just putting a load of dial/sip extensions on a line? :P |
| 11:45.05 | *** join/#asterisk iamy_china (~iamy_chin@221.223.51.31) |
| 11:46.30 | *** join/#asterisk Deathvalley122 (~Death@unaffiliated/deathvalley122) |
| 11:46.52 | Deathvalley122 | hello asterisk is open source right? |
| 11:46.53 | Chainsaw | E-bola: If I read that right, your soundstation is not sending a user name. |
| 11:46.55 | Chainsaw | Deathvalley122: Yes :) |
| 11:46.56 | beardy | I guess that is what happens, jut aliased/shortened syntax. |
| 11:47.09 | *** join/#asterisk garymc (~chatzilla@host81-148-29-236.in-addr.btopenworld.com) |
| 11:47.18 | Deathvalley122 | how come I see a list of prices for asterisk? |
| 11:47.35 | Chainsaw | Deathvalley122: Because you may want a commercially supported version of Asterisk, rather than an open source version. |
| 11:47.50 | Chainsaw | Deathvalley122: If you are a business that does not employ any programmers for example, the availability of source code may not sway you. |
| 11:47.52 | Deathvalley122 | ohh commerical I see |
| 11:47.56 | beardy | Because you're looking at digium.com's priced products? |
| 11:48.07 | E-bola | Chainsaw: Im pretty sure its indeed the soundstation thats the issue, problem is i got no idea how to fix it |
| 11:48.17 | E-bola | After changing the password on asterisk it started doing this :( |
| 11:48.18 | Deathvalley122 | what are the list of 1-800 numbers? |
| 11:48.33 | Chainsaw | E-bola: I have a SoundStation 670 here, authenticating to 1.6.2.9 successfully. |
| 11:48.34 | Deathvalley122 | or any numbers for that matter |
| 11:48.52 | E-bola | Chainsaw: cant u set the ip address on the web interface on that one either? |
| 11:48.54 | Chainsaw | E-bola: How are you provisioning them? Automatically over HTTP? Or are you manually setting things? |
| 11:48.57 | E-bola | The firmware seems a bit odd to me |
| 11:49.07 | E-bola | Chainsaw: only have 1 polycom, so everything manualy |
| 11:49.12 | Chainsaw | E-bola: I use DHCP and auto-provision over HTTP. |
| 11:49.16 | drmessano | Deathvalley122: Huh? |
| 11:49.27 | E-bola | Ya if i read the admin guide correctly, that allows you to control more stuff |
| 11:49.45 | Deathvalley122 | the 1-800 numbers you get with asterisk |
| 11:49.48 | Chainsaw | E-bola: Now it's asking me for a password that I don't know. Hah. |
| 11:49.52 | Chainsaw | E-bola: Never done it manually. |
| 11:49.52 | Deathvalley122 | what are the list of them |
| 11:50.04 | drmessano | They're on the website |
| 11:50.15 | Deathvalley122 | where? |
| 11:50.39 | redax | hi.. |
| 11:50.47 | drmessano | "Contact us" |
| 11:51.16 | redax | is there a way to get app_voicemail as dumb as listening messages / next/prev and delete... so no folder options no greeting recordings etcetc... |
| 11:53.00 | beardy | redax: Grab it, rewrite. |
| 11:53.13 | Chainsaw | redax: I do not see suitably options on VoiceMailMain, no. |
| 11:53.37 | Chainsaw | redax: So you will have to implement most of it yourself. (Doable, but no silver bullet) |
| 11:54.58 | redax | aha. |
| 11:55.24 | redax | is app_minivm useable for a minimalistic stuff? |
| 11:55.59 | redax | I've found a few documentation about minivm, but first I tried app_voicemail. |
| 11:56.00 | tzafrir_laptop | Deathvalley122, open source may still be commercially supported |
| 11:56.11 | redax | but seems like app_voicemail kinda hardcoded... |
| 11:57.11 | tzafrir_laptop | I bet Chainsaw would be willing to provide you commercial support for Asterisk, for the right price ;-) |
| 11:58.14 | tzafrir_laptop | (or maybe also provided it's on a Linux system of a specific distribution. I heard those who provide commercial support have wierd requirements) |
| 11:58.44 | Deathvalley122 | lol |
| 11:59.02 | Deathvalley122 | commerical seems a bit out of my price range tbh |
| 11:59.57 | *** join/#asterisk bn-7bc (bjarne@mac.wlan.noare-1.holmedal.net) |
| 12:00.01 | hmodes | and so you find yourself here, dealing with us assholes |
| 12:00.09 | redax | huh. nobody tried minivm ? :-o |
| 12:08.44 | *** join/#asterisk n3hxs (~HAMming@ip67-88-206-99.z206-88-67.customer.algx.net) |
| 12:10.34 | tzafrir_laptop | redax, hmm... yes, app_voicemail is hardcoded. Have you tried minivm? |
| 12:11.02 | redax | not yet tzafrir, just reading that small documentation of it |
| 12:11.16 | redax | seems to be sufficient for my needs for the first sight |
| 12:14.04 | Deathvalley122 | eh I dunno the commerical packages seem expensive to me ... |
| 12:14.06 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 12:14.25 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:15.31 | Chainsaw | Deathvalley122: The commercial support basically replaces a full-time position at your company. |
| 12:15.46 | Chainsaw | Deathvalley122: And it is priced as such. |
| 12:16.07 | Deathvalley122 | 500 dollars is too much Chainsaw ... |
| 12:16.16 | Deathvalley122 | I can not afford that |
| 12:16.43 | Chainsaw | Deathvalley122: Okay, so consultancy & commercial support is out. Asterisk can still be used; the download is free. |
| 12:16.57 | Chainsaw | Deathvalley122: But you will be doing the work yourself. I hope you understand that. Writing a dial-plan, etc. |
| 12:17.28 | Deathvalley122 | not really I am new to this Voip stuff ... |
| 12:17.44 | [TK]D-Fender | ~book |
| 12:17.45 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 12:17.46 | [TK]D-Fender | ^^^^^^^^^^^ |
| 12:17.52 | Chainsaw | Indeed Fender, I second that suggestion. |
| 12:18.07 | *** join/#asterisk aidinb (~Aidin@71-95-223-217.dhcp.mtpk.ca.charter.com) |
| 12:18.11 | beardy | So was I a couple of months ago. Hppy reading. |
| 12:18.18 | beardy | Happy* |
| 12:23.57 | tzafrir_laptop | well, if Digium's prices are high, there's a business oportunity for someone else to provide a cheaper support ;-) |
| 12:24.46 | Chainsaw | I limit the time I spend on Asterisk. It's better for my blood pressure. |
| 12:25.13 | beardy | I do the same but with people. |
| 12:26.04 | Chainsaw | People are usually fine. It's the amount of shouting I have to do before people look at patches here. |
| 12:28.36 | beardy | :) |
| 12:34.38 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 12:34.56 | Deathvalley122 | Chainsaw: where can I buy a 1-800 number or does asterisk provide 1-800 numbers to use to contact your guys support for me to use if that makes sense |
| 12:35.23 | Chainsaw | Deathvalley122: Digium is not a telco. |
| 12:35.45 | [TK]D-Fender | Deathvalley122: Asterisk does not provide "services". It lets you USE them however |
| 12:35.56 | *** join/#asterisk oej (~olle@ns.webway.se) |
| 12:35.57 | Chainsaw | I see Fender has this in hand. Lunchtime for me. |
| 12:35.58 | [TK]D-Fender | Deathvalley122: .... |
| 12:36.00 | [TK]D-Fender | ~itsp |
| 12:36.01 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is 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. |
| 12:36.02 | [TK]D-Fender | ^^^ |
| 12:36.18 | Deathvalley122 | so how is my clients gonna call us if I set this up? |
| 12:36.33 | [TK]D-Fender | Deathvalley122: Depensd hoy you CHOOSE to set it up |
| 12:36.41 | [TK]D-Fender | Deathvalley122: Depends how you CHOOSE to set it up |
| 12:37.07 | [TK]D-Fender | Deathvalley122: * processes calls. How you arrange to have them arrive is up to you. |
| 12:37.24 | Deathvalley122 | I want to use a 1-800 number so that the clients can call us |
| 12:37.30 | [TK]D-Fender | Deathvalley122: Analog lines, digital trunks, VoIP protocol of some sort, etc... all up to you |
| 12:37.30 | Deathvalley122 | instead of using the ip |
| 12:37.54 | [TK]D-Fender | Deathvalley122: Again, the # is a PSTN #. How it physically arrives at your server is up to you |
| 12:38.29 | [TK]D-Fender | Deathvalley122: Traditional telco's off physical lines to deliver calls, you are you use an ITSP, etc |
| 12:38.40 | [TK]D-Fender | [08:35]<infobot>[~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is 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. |
| 12:38.42 | [TK]D-Fender | ^^^ |
| 12:39.06 | Deathvalley122 | thing is this is goning be a rented dedicated server the hardware of it is out of my hands |
| 12:39.20 | beardy | Deathvalley122: Asterisk is a PBX, a telephony switch implemented in software. What kind of telephony services, technologies, "phone numbers" and so on you want to use, is up to you. |
| 12:39.27 | *** join/#asterisk garymc (~chatzilla@host81-148-29-236.in-addr.btopenworld.com) |
| 12:40.24 | Deathvalley122 | hmm |
| 12:41.29 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 12:42.05 | *** join/#asterisk guilhermebr (~Guilherme@200.103.96.98) |
| 12:42.23 | Deathvalley122 | is confused |
| 12:43.24 | [TK]D-Fender | Deathvalley122: What is confusing about this? |
| 12:43.47 | beardy | Deathvalley122: To hook up a DID (phone number) of any kind, you buy a SIP trunk from an ITSP (see above), cnnect to it, and have the calls to your purchased/rented number be sent to your Asterisk, and handled in the way you cnfigure it to. |
| 12:44.30 | Deathvalley122 | I dunno where to get one |
| 12:44.33 | beardy | sprinkles some o:s |
| 12:44.54 | [TK]D-Fender | Deathvalley122: maybe if you read the blurb I had infobot give you TWICE <- |
| 12:45.07 | Deathvalley122 | yea no links |
| 12:45.10 | Deathvalley122 | I need links |
| 12:45.11 | Deathvalley122 | lol |
| 12:45.11 | [TK]D-Fender | Deathvalley122: Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. <-------------------- |
| 12:45.19 | [TK]D-Fender | ENTER FOR A LISTING |
| 12:46.00 | [TK]D-Fender | waits to see some minimal neuro-electric activity |
| 12:46.03 | beardy | It could say "/msg infobot itsplist-us" to be more clear. |
| 12:46.16 | *** join/#asterisk pabelanger (~pabelange@yoda.kanatek.com) |
| 12:46.17 | [TK]D-Fender | beardy: No need.... it sya it even more direct and shorter |
| 12:46.20 | [TK]D-Fender | says* |
| 12:46.34 | beardy | "Enter" where? one wonders. |
| 12:46.54 | beardy | Some surely do anyway. |
| 12:47.36 | TJNII | Well, most people see someone enter ~itsp and put 2 and 2 together. |
| 12:47.58 | [TK]D-Fender | beardy: Yeah, and how was that blurb "activated" exactly? It not like we haven't spammed up half a dozen of these since his arrival |
| 12:50.15 | Deathvalley122 | okay |
| 12:50.16 | Deathvalley122 | I see |
| 12:51.22 | [TK]D-Fender | Deathvalley122: Show us :) |
| 12:51.42 | Deathvalley122 | its kinda like a domain almost |
| 12:51.43 | Deathvalley122 | lol |
| 12:51.52 | garymc | Hi im reading that the c option for dial commands for 1.6.2 is in the app_queue file. Anyone know where that file is located? |
| 12:53.16 | garymc | sorry that should been in freepbx channel ^ |
| 12:53.26 | [TK]D-Fender | garymc: CORRECT |
| 12:53.40 | garymc | what is correct? |
| 12:54.24 | garymc | freepbx one of the file name? or both ? :P |
| 12:54.29 | beardy | Your retraction I suppose. |
| 12:54.34 | garymc | oh |
| 12:56.30 | Deathvalley122 | [TK]D-Fender: sorry for the frustrations just kinda new to all of this. |
| 12:56.36 | beardy | Deathvalley122: Yes. It has many similarities. |
| 12:57.18 | joobie | fark |
| 12:57.23 | joobie | im out of red wine :/ |
| 12:58.21 | TJNII | Breakfast of Champions. |
| 12:58.42 | beardy | ges to move cows. |
| 12:58.46 | beardy | goes |
| 12:58.58 | garymc | drinking wine at this time |
| 12:59.17 | Deathvalley122 | wow talk about a long doc you guys have ... |
| 13:00.54 | *** join/#asterisk grEvenX (~even@apb9hb.ip.ssc.net) |
| 13:02.04 | drmessano | Deathvalley122: Asterisk is a full out telephony platform used for creating anything from a simple voicemail system, to a very complex PBX system.. Learning Asterisk is like learning a new programming language |
| 13:02.10 | [TK]D-Fender | Deathvalley122: What did that blurb tell you do? |
| 13:02.51 | Deathvalley122 | it gave me a list of links |
| 13:03.18 | joobie | garymc, 11PM here |
| 13:03.20 | *** join/#asterisk cdahmedeh (~cdahmedeh@62.68.65.246) |
| 13:03.26 | Deathvalley122 | where I can rent itsp |
| 13:03.30 | garymc | ahhh nice |
| 13:03.38 | Deathvalley122 | in the U.S. |
| 13:03.51 | joobie | boys |
| 13:03.53 | joobie | im in AU |
| 13:03.58 | tzafrir_laptop | infobot, tell Deathvalley122 about itspus |
| 13:04.04 | [TK]D-Fender | Deathvalley122: What did that blurb tell you do? <------------ |
| 13:04.12 | joobie | anyone want to hook up some international IAX's to benefit on call oruting |
| 13:04.16 | [TK]D-Fender | [08:45]<[TK]D-Fender>Deathvalley122: Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. <-------------------- |
| 13:04.19 | joobie | i can pass to the AU network for free |
| 13:04.21 | joobie | for landline |
| 13:04.24 | tzafrir_laptop | infobot, tell Deathvalley122 about itsplist-us |
| 13:04.31 | Deathvalley122 | yea I did it in msg |
| 13:04.44 | joobie | anyway want to setup some trunks for some cheap routing? |
| 13:04.53 | joobie | -anyway+anyone |
| 13:05.03 | joobie | free AU calls! :P |
| 13:05.10 | joobie | .. just not mobile |
| 13:05.19 | Deathvalley122 | I think I found a cheap one I can actually get a number locally ... |
| 13:06.06 | [TK]D-Fender | Deathvalley122: 1-800 has nothing to do about "local" really.... |
| 13:06.44 | Deathvalley122 | no it doesnt but I found one locally from a site |
| 13:07.08 | Deathvalley122 | its not a 1-800 number either |
| 13:07.30 | *** join/#asterisk t_dot_zilla (~chatzilla@firewall-a.buf.ny.i-evolve.net) |
| 13:09.06 | *** join/#asterisk patrick^ (~patrick_@2001:470:1d:349:219:21ff:fe4e:f5de) |
| 13:10.13 | Deathvalley122 | okay |
| 13:10.42 | [sr] | people |
| 13:10.56 | [sr] | it's possible to block an incoming call that doesn't have a CID? |
| 13:11.04 | [sr] | for now i just want to know it its possible |
| 13:11.16 | [TK]D-Fender | [sr]: You can do whatever you want with your calls |
| 13:11.32 | Deathvalley122 | well when I have time I will sit down for a certain hours and read bits and pieces of the doc not gonna have time to read it all |
| 13:11.34 | Deathvalley122 | lol |
| 13:11.35 | [sr] | block it, and play a recoded message, can it be done? |
| 13:11.43 | [TK]D-Fender | Deathvalley122: All of those ITSPs offer Toll-free services |
| 13:12.03 | [TK]D-Fender | [sr]: What does "block it" mean when you are answering the call and playing thema message? |
| 13:12.12 | [sr] | [TK]D-Fender: that! |
| 13:12.28 | [sr] | not exactly block, answer and play the message |
| 13:12.46 | Deathvalley122 | http://www.voicepulse.com/connect/ not that one its all area code ones [TK]D-Fender |
| 13:12.54 | joobie | [sr], why do u want to do that? |
| 13:13.02 | [sr] | joobie: just curious :) |
| 13:13.24 | [TK]D-Fender | Deathvalley122: Right hand column on THAT page ---->Local & Toll-Free Numbers |
| 13:13.29 | [TK]D-Fender | Deathvalley122: TOLL FREE |
| 13:16.51 | Deathvalley122 | ah |
| 13:17.42 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:17.42 | *** mode/#asterisk [+o malcolmd] by ChanServ |
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| 13:29.04 | *** part/#asterisk rossand (~aross@CPE00222ddf42e0-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 13:29.40 | pabelanger | Q: ACLs with domains names? Does that make sense? |
| 13:31.45 | t_dot_zilla | we are having a problem with a line staying active after it's completed a transfer....has anybody dealt with an issue like this....is there a setting that needs to be changed in asterisk? |
| 13:31.50 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 13:31.50 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:34.49 | *** join/#asterisk andresm (~andresm@ubuntu/member/andresmujica) |
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| 13:37.49 | [TK]D-Fender | t_dot_zilla: Perhaps you could validate your scenario for us... |
| 13:38.42 | t_dot_zilla | actually i cannot...for some reason it is only happening at one customer location and i am unable to go to the location |
| 13:39.15 | hrhrhr | putnopvut: is masking available to everyone? |
| 13:40.07 | putnopvut | masking? |
| 13:40.15 | hrhrhr | mode +x |
| 13:40.19 | hrhrhr | or whatever it's called on here |
| 13:42.03 | *** join/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 13:42.47 | *** part/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
| 13:43.30 | *** join/#asterisk ajm0716 (~ajm0716@c-76-97-74-68.hsd1.ga.comcast.net) |
| 13:43.43 | [TK]D-Fender | t_dot_zilla: SSH doesn't care where you are, and that has nothing to do with even describing the precise circumstances involved |
| 13:45.01 | t_dot_zilla | [TK]D-Fender: the servers are here, i can access asterisk and do a sip trace the problem is, there are 100s of calls going on every second and i will be unable to locate the sip trace for that customer |
| 13:46.07 | [TK]D-Fender | t_dot_zilla: You shouldn't ask us to fix a problem you can't show us.... And that doesn't stop you from doing the other part... |
| 13:46.21 | *** join/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 13:47.09 | t_dot_zilla | the circumstances are.....phone in queue recieves call, call is transfered to another phone outside queue, phone in queue has active line until transfered call has ended, during this time the line is supposedly active and will not recieve incoming calls but does have a dialtone |
| 13:47.15 | leifmadsen | hrhrhr: what does +x do? |
| 13:47.23 | leifmadsen | hrhrhr: likely only ops can set masks on the channel |
| 13:47.44 | hrhrhr | mode +x is hostname masking |
| 13:47.56 | hrhrhr | it's normally accessible via some sort of reg |
| 13:48.00 | hrhrhr | ie nickserv |
| 13:48.04 | hrhrhr | but i dont see the option on freenode |
| 13:48.11 | hrhrhr | clearly, it is available tho |
| 13:49.03 | leifmadsen | hrhrhr: I think you have to speak with freenode staff to change it |
| 13:49.10 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 13:49.26 | leifmadsen | typically people in prominent projects or people who donate get hostmasks |
| 13:49.37 | leifmadsen | I'm not sure what the rules are though |
| 13:50.37 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
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| 14:00.39 | t_dot_zilla | we are having the same problem described here....http://fonality.com/trixbox/forums/trixbox-pro/hudlite-hud-pro-trixbox-pro/call-transferred-my-extension-call-still-shown-active |
| 14:03.40 | *** join/#asterisk jmacz (~jmacz@190.144.75.22) |
| 14:04.08 | leifmadsen | reading through that forum I'm not convinced there is a bug present. Anyone who says they have it haven't been able to provide enough information to reproduce it on a consistent basis or really explain what is going on. |
| 14:05.45 | *** join/#asterisk Khratos (~jespinal@66.128.60.148) |
| 14:05.53 | garymc | Ok i think this is the right place. I need to get my reason headers working. How do I start. my phones firmware are upto date and my polycoms apparently support it. But I cant get it working |
| 14:05.57 | garymc | anyone help me out |
| 14:06.10 | *** join/#asterisk eliel (~eliels@201.234.94.226) |
| 14:06.14 | Khratos | Is there some 'setting' that could lead dahdi installation not to create the /etc/dahdi directory? |
| 14:07.39 | t_dot_zilla | leifmadsen: why would someone lie about a problem that is occuring? |
| 14:07.52 | leifmadsen | t_dot_zilla: I didn't say they were lieing about the problem |
| 14:08.04 | *** join/#asterisk Ryushin (proxy@windwalker.openinnovations.com) |
| 14:08.23 | leifmadsen | t_dot_zilla: what I said was that I'm not convinced it is actually a bug (could be a configuration issue) because no one has been able to provide information on how to reproduce it consistently |
| 14:08.59 | t_dot_zilla | leifmadsen: it could be a configuration issue....but what config would cause that problem to occur? |
| 14:09.32 | leifmadsen | t_dot_zilla: if I knew that, then I'd have already replied on the forum stating what the issue is -- I'm saying not enough information has been provided in order to diagnose what the real problem is |
| 14:10.23 | t_dot_zilla | leifmadsen: it is difficult to know what is going on because i have phones on the same asterisk box with identical setup and i cannot replicate the problem |
| 14:10.42 | leifmadsen | t_dot_zilla: and either can I, so we're at the same spot we were several minutes ago |
| 14:11.11 | t_dot_zilla | leifmadsen: but that kind of eliminates a configuration problem because i'm using the same asterisk box that the customer is |
| 14:11.32 | leifmadsen | yes |
| 14:11.55 | *** join/#asterisk RobH (~robh@wikimedia/RobH) |
| 14:24.22 | [TK]D-Fender | [09:47]<t_dot_zilla>the circumstances are.....phone in queue recieves call, call is transfered to another phone outside queue, phone in queue has active line until transfered call has ended, during this time the line is supposedly active and will not recieve incoming calls but does have a dialtone <-- what PHONES? Transfered HOW? |
| 14:25.21 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 14:25.57 | t_dot_zilla | [TK]D-Fender: the phone (polycom) that transfered the call has a dialtone yet will not receive incoming calls....this happens if the call is blind / normal transfer using the polycom softkeys |
| 14:26.31 | [TK]D-Fender | t_dot_zilla: Because you need to use * based DTMF transfers otehrwise the queue will not know that they aren't on the same call. |
| 14:26.37 | [TK]D-Fender | t_dot_zilla: because of how it is handed off. |
| 14:27.24 | t_dot_zilla | [TK]D-Fender: so you are saying use the # (or whatever you have set in * to do a xfer) ? |
| 14:27.39 | [TK]D-Fender | t_dot_zilla: Yes |
| 14:27.41 | *** join/#asterisk ruyo (~psantos@195.23.253.223) |
| 14:27.50 | t_dot_zilla | ok |
| 14:40.28 | *** join/#asterisk grEvenX (~even@ti0057a380-2069.bb.online.no) |
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| 14:49.56 | *** part/#asterisk c0rnoTa (~c0rnoTa@nas-8.emserv.ru) |
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| 14:54.35 | hmodes | I was not paying attention, come again? Should we use phaetons? |
| 14:55.00 | *** join/#asterisk Tagor (~none@s55928c6d.adsl.wanadoo.nl) |
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| 14:55.17 | Tagor | Is there a way to Dial() and then Playback() when the phone is picked up? |
| 14:55.23 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-32-170-204.mia.bellsouth.net) |
| 14:55.30 | Tagor | It doesn't continu with the Playback() after the Dial() |
| 14:56.01 | hmodes | I'm pretty sure this was covered earlier, you either answer() or you don't |
| 14:56.02 | Katty | so. |
| 14:56.06 | Katty | i got to work this morning. |
| 14:56.08 | Chainsaw | Hi Katty :) |
| 14:56.11 | Katty | and at 9 i got tired. |
| 14:56.26 | Chainsaw | Katty: This is a Monday thing. I'm fairly sure it happens to all of us. |
| 14:56.27 | Katty | and i just jolted awake in my chair. |
| 14:56.33 | [TK]D-Fender | Tagor: "core show application dial" <----------- |
| 14:56.34 | Katty | (an hour later) |
| 14:56.34 | Chainsaw | Katty: Were you spotted? |
| 14:56.42 | Katty | Chainsaw: not that i know of :P |
| 14:56.48 | Katty | Chainsaw: if someone did walk in, they didn't wake me up. |
| 14:57.00 | Tagor | hmodes: I have this: s,1,Answer() s,2,Dial(SIP/xxxx@default) s,3,Playback(something) but it won't playback during the dial() |
| 14:57.03 | Chainsaw | Katty: Coffee time :D |
| 14:57.08 | Katty | Chainsaw: eeek, no |
| 14:57.27 | Chainsaw | Katty: Ehm, tea time? It doesn't work as well... |
| 14:57.33 | hmodes | Well then, you answered |
| 14:57.33 | Katty | Chainsaw: my stomach is a bit messed up. had a few sips of sprite this morning and started having problems again |
| 14:57.43 | Tagor | [TK]D-Fender: I tried 'g' but that only playbacks when the other party hangs up |
| 14:57.44 | Chainsaw | Katty: Ah, quite. |
| 14:57.54 | hmodes | either answer the call, or don't. It's as simple as that. |
| 14:57.56 | Chainsaw | Katty: Loud music? |
| 14:58.01 | Katty | Chainsaw: been having stomach cramps, which radiates real bad and makes it feel like my chest hurts |
| 14:58.18 | Tagor | hmodes: that's what I did |
| 14:58.22 | Chainsaw | Katty: That sucks, I'm sorry :( |
| 14:58.24 | Katty | Chainsaw: oh i'm wide awake now :> |
| 14:58.41 | [TK]D-Fender | Tagor: Clearly "g" isn't the right answer <- |
| 14:58.47 | Katty | Chainsaw: all good (= they're getting better. |
| 14:58.58 | hmodes | Okay, so you answered() it and then dialed elsewhere |
| 14:59.12 | hmodes | am I not following your logic? |
| 14:59.39 | *** join/#asterisk wonderworld (~ww@ip-62-143-22-226.unitymediagroup.de) |
| 15:00.02 | hmodes | you answered the call, you'll be billed for that |
| 15:00.28 | *** join/#asterisk waschtl (~waschtl@3ed8a589.d.d9tcloud.de) |
| 15:00.31 | hmodes | that is tantamount to a session progress |
| 15:00.58 | hmodes | there's no getting around this |
| 15:01.05 | [TK]D-Fender | [10:56]<Tagor>hmodes: I have this: s,1,Answer() s,2,Dial(SIP/xxxx@default) s,3,Playback(something) but it won't playback during the dial() <- of course not. When you hit priority 3, you have LEFT dial, and your call is ENDED.. or never happened |
| 15:01.10 | *** join/#asterisk bmoraca_work (~bmoraca@66-242-174-254.ceres.bvn.net) |
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| 15:01.50 | hmodes | shit, seriously? |
| 15:02.23 | hmodes | I fail, it seems |
| 15:02.31 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 15:03.22 | beardy | Yes. I wonder though, if the other end hangs up, then you might come back to the dialplan. But you can never trust that anything after Dia() gets run. |
| 15:03.59 | hmodes | I should stop being authoritative, clearly. |
| 15:04.12 | Tagor | [TK]D-Fender: thanks, I didn't saw the a-option. But this doesn't work: s,2,Dial(SIP/xxxxxxxx@default,20,a(/var/lib/asterisk/sounds/something)) it does dial, but doesn't play the file |
| 15:04.47 | beardy | hmodes: If you Background() something before the Dial(), it might go on playing whie dialing.. experiment and report. |
| 15:05.36 | *** join/#asterisk LemensTS (~LemensTS@adsl-70-238-159-189.dsl.stlsmo.sbcglobal.net) |
| 15:05.47 | [TK]D-Fender | nope |
| 15:05.58 | [TK]D-Fender | Tagor: I'm not seeing you showing the call and your backup |
| 15:06.10 | hmodes | I don't spend nearly enough time with agis, but I'll try that and mebbe demand some moosepenis... |
| 15:06.54 | Tagor | [TK]D-Fender: what do you mean with the backup? The only thing I have is: s,1,anwer() s,2,Dial(see-above) s,3,hangup() |
| 15:07.00 | LemensTS | i installed 1.6.2.9 from source, wanting to go to 1.6.2.10. do i just download teh src of 2.10, then ./configure, make, make install like i did when i installed 1.6.2.9 (without doing make samples of course) |
| 15:07.28 | [TK]D-Fender | Tagor: Look. At. The. CALL |
| 15:07.32 | [TK]D-Fender | tageand the FILES. |
| 15:07.38 | [TK]D-Fender | Tagor: and the FILES. |
| 15:07.40 | beardy | LemensTS: The README and/or INSTALL file will say. |
| 15:07.52 | [TK]D-Fender | LemensTS: Yes |
| 15:07.55 | hmodes | what does your code DO??? |
| 15:08.29 | LemensTS | beardy: ok i was reading the upgrade.txt files, that was the problem then. :) thanks guys |
| 15:12.42 | Tagor | [TK]D-Fender: Ok, I got it working now, thanks. But it won't pass audio from and to both parties when playing the file (the persons can't talk). Is there a way to talk while playing the file? |
| 15:13.02 | [TK]D-Fender | Tagor: Not by this means |
| 15:13.44 | Tagor | [TK]D-Fender: Is there a way other then setting up a conference? |
| 15:15.06 | [TK]D-Fender | Tagor: Originate 2 local chanspy+whisper channels to taget teh 2 legs of the call |
| 15:17.19 | *** part/#asterisk LemensTS (~LemensTS@adsl-70-238-159-189.dsl.stlsmo.sbcglobal.net) |
| 15:27.34 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 15:30.05 | ruyo | How can I force a channel hangup? (soft hangup doesn't hangup) |
| 15:30.54 | [TK]D-Fender | ruyo: Show us. |
| 15:31.41 | *** join/#asterisk outtolunc (~me@adsl-66-218-53-172.dslextreme.com) |
| 15:31.49 | ruyo | Ok, sec. |
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| 15:37.18 | ruyo | [TK]D-Fender, http://pastebin.com/UGXiiYNb |
| 15:37.34 | ruyo | Channels keep getting stuck I don't know why. |
| 15:38.00 | ruyo | But I'd like to be able to manually hangup them before I check what's going on. |
| 15:40.08 | leifmadsen | hmodes: try something like the M() or U() options to DialI() -- may do what you need |
| 15:42.10 | pabelanger | Q: When asterisk does transcoding of g711 to g729, is the process in asterisk g711 -> sln -> g729? |
| 15:42.21 | russellb | A: Yes. |
| 15:42.23 | [TK]D-Fender | ruyo: Ok.... use AMI Redirect to drop them off a cliff <- |
| 15:42.25 | hmodes | I never pay attention to the options. I either answer, or I don't. |
| 15:43.36 | ruyo | [TK]D-Fender, ok, I'll try that. |
| 15:44.14 | ruyo | By the way, can this be the result of a bad dialplan? |
| 15:44.28 | *** part/#asterisk asteriskATmarmuD (~Mundt@193.158.65.23) |
| 15:44.49 | ruyo | The CPU is at 100% most of the time. |
| 15:44.56 | hmodes | it can always be the result of a bad dialplan |
| 15:45.04 | pabelanger | russellb: thanks |
| 15:45.06 | beardy | Any diallan that doesn't meet your requirement is bad by definition, isn't it? :) |
| 15:45.09 | Chainsaw | ruyo: It could also be heavy transcoding, which could be due to a bad sip.conf |
| 15:45.10 | ruyo | It's not a loop though. |
| 15:46.10 | ruyo | There is no transcoding, everything is alaw. |
| 15:46.33 | *** join/#asterisk pabelanger (~pabelange@yoda.kanatek.com) |
| 15:46.34 | ruyo | With version 1.4.28 I had segfaults, this is .34 |
| 15:48.19 | *** join/#asterisk datacompboy (~datacomp@l49-3-84.cn.ru) |
| 15:48.32 | datacompboy | Hi! Everybody knows where I can find say.conf for slovenian language? |
| 15:49.11 | *** join/#asterisk Godfather_ (~Godfather@50.Red-79-155-185.dynamicIP.rima-tde.net) |
| 15:51.39 | Godfather_ | hi |
| 15:51.50 | Godfather_ | how can i enable a sip set debug on a specify user? |
| 15:52.15 | [TK]D-Fender | Godfather_: "help sip" <-- |
| 15:52.56 | hmodes | bacon solves all problems. |
| 15:53.20 | *** join/#asterisk raden_work (~jon@69-179-99-17.stat.centurytel.net) |
| 15:53.44 | Godfather_ | [TK]D-Fender, but this peer is uneable to log in |
| 15:53.54 | [TK]D-Fender | Godfather_: "help sip" <-- |
| 15:53.59 | Godfather_ | then it says me Unable to get IP address of peer '100' |
| 15:54.12 | [TK]D-Fender | Godfather_: then SPECIFY THE IP TO LOOKS FOR |
| 15:54.22 | Godfather_ | nice idea |
| 15:56.46 | Kobaz | that's weird |
| 15:57.31 | Kobaz | t1 circuit is up... no pri signalling on it |
| 15:58.56 | Godfather_ | i trying to log in my asterisk. I'm outside the asterisk lan. I configured externip and localnet, but the telephone doesnt register |
| 15:59.16 | *** join/#asterisk Sipster_ (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 15:59.29 | Godfather_ | [TK]D-Fender, i pastebin you some stuff if you can see something wrong.. |
| 15:59.53 | hmodes | it's not so strange, if the ip doesn't register, yell. |
| 16:00.18 | Godfather_ | i recieve the packets and i configure the peer with nat=yes |
| 16:00.33 | beardy | Godfather_: Is the NAT:ing firewall forwarding the ports you run asterisk on. (SIP, 5060 by default)? |
| 16:00.47 | Godfather_ | beardy, yes, i opened 5060 and 10k to 20k |
| 16:01.08 | hmodes | nat=grape |
| 16:01.21 | [TK]D-Fender | Godfather_: Look at GLOBAL SIP debug |
| 16:01.33 | beardy | Godfather_: Forwarded too, not only opening? |
| 16:01.44 | Godfather_ | [TK]D-Fender, ok, 1 sec |
| 16:02.05 | Godfather_ | beardy, yes forwarded. |
| 16:03.30 | *** join/#asterisk Z_God (~julius@wlan234115.mobiel.utwente.nl) |
| 16:07.01 | Godfather_ | [TK]D-Fender, do you see anything? |
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| 17:04.14 | beardy | Not very nice to quit when one spends almost an hour or more trying to help.. |
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| 17:05.22 | Qwell | beardy: welcome to IRC. |
| 17:06.07 | *** join/#asterisk keith4 (~keith@unaffiliated/keith4) |
| 17:06.08 | beardy | I've been here many years, and have had that countless times. |
| 17:06.30 | Qwell | then surely you aren't still surprised by it? |
| 17:06.38 | beardy | Exercising whineright. |
| 17:06.49 | *** join/#asterisk uqlev (~yuriy@91.184.221.31) |
| 17:07.41 | beardy | I give humans the benefit of the doubt from time to time. :) |
| 17:07.51 | Qwell | users != human |
| 17:08.47 | beardy | I'm well reminded. |
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| 18:14.30 | mcr | using 1.4, is there a setting in sip.conf that controls how often the SIP REGISTER is sent? Default seems to be 120s, but I want to change it to 60s to help debug some things. |
| 18:15.31 | mcr | I found, defaultexpirey at http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf, and I'm guessing it's typo for defaultexpiry. |
| 18:17.05 | [TK]D-Fender | mcr: Correct. The WIKI is often wrong |
| 18:17.15 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 18:17.24 | wr| | can anyone tell me if it's possible to use mISDN V2 w/ asterisk? installing misdn doesn't enable the chan_misdn option in menuselect, that's why I'm asking. |
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| 18:25.39 | Godfather_ | [TK]D-Fender i didnt solve the problem, can you try to register to my server..? |
| 18:26.41 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 18:28.29 | [TK]D-Fender | Godfather_: Not today, sorry. |
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| 18:35.25 | fauxalliance | wr|, ugg,, is that the domain of the 'Linux Call Router' |
| 18:36.56 | wr| | fauxalliance, I don't understand the question. but as I couldn't get misdn directly to work I proceeded with trying to use LCR (which fails to build, just sent mail to their list) |
| 18:37.10 | wr| | you don't know by chance which asterisk version LCR 1.7 expects? |
| 18:37.21 | wr| | all failures there came from ast_* calls... |
| 18:37.51 | WIMPy | I've used it with both 1.4 and 1.6. |
| 18:38.34 | WIMPy | chan_misdn only works with misdn1 which is no longer supported since 10 kernel version or something. |
| 18:38.45 | WIMPy | lcr needs misdn2 from a current kernel. |
| 18:40.28 | wr| | lcr says: chan_lcr.c:2382: error: dereferencing pointer to incomplete type |
| 18:41.36 | WIMPy | Current kernel? Curent version of misdn_user? |
| 18:43.46 | raden_work | is there a way to set a passcode to dial out ? |
| 18:44.03 | raden_work | so like anyone can dial anyone in building but needs a code to dial a outside line ? |
| 18:44.26 | Katty | ohai |
| 18:44.33 | raden_work | so if i dial 15555555555 it will prompt for a password |
| 18:44.39 | raden_work | then enter a 4 digit code or something |
| 18:45.06 | [TK]D-Fender | raden_work: Your dialplan does whatever you tell it to |
| 18:45.37 | raden_work | can it make me toast ? |
| 18:45.38 | raden_work | :) |
| 18:46.00 | [TK]D-Fender | raden_work: Yes. Mine did this for me years ago. |
| 18:46.06 | raden_work | LMAO |
| 18:46.09 | [TK]D-Fender | raden_work: And no, I'm not kidding. |
| 18:46.34 | raden_work | I bet anything is possible with proper interfaces |
| 18:48.05 | raden_work | I love asterisk, so many possibilities |
| 18:53.05 | *** part/#asterisk mcr (~mcr@2001:4830:16ca:1:20d:60ff:fefa:7f03) |
| 18:54.22 | *** join/#asterisk [netman] (~netman@83.54.228.88) |
| 18:54.37 | wr| | WIMPy, linux 2.6.34, misdn 2.0.1 |
| 18:54.53 | wr| | asterisk 1.6.2.6 |
| 18:55.29 | WIMPy | misdn? That's in the kernel. |
| 18:55.47 | wr| | misdnuser I mean of cause :) |
| 18:57.16 | Katty | pokes at raden_work |
| 18:57.27 | WIMPy | That's what I use as well, but Asterisk 1.6.2.9 |
| 18:57.29 | *** join/#asterisk AlHafoudh (~AlHafoudh@adsl-dyn129.78-98-235.t-com.sk) |
| 18:57.42 | wr| | I'll try that. |
| 19:01.13 | *** join/#asterisk QubeZ (~nkasu@64.128.254.34) |
| 19:01.15 | QubeZ | hello all |
| 19:01.23 | raden_work | pokes at Katty |
| 19:02.00 | QubeZ | anyone here compiled the wanpipe-voicetime drivers for UT50 usb voice synch tool? I'm wondering if it is supposed to show up as a DADHI device because when I run dahdi_hardware, it doesnt show up. However, the timing is great and I see its being used via cat /proc/dahdi/1 |
| 19:04.47 | *** join/#asterisk n3hxs (~HAMming@wsip-68-15-181-101.ph.ph.cox.net) |
| 19:11.36 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 19:14.27 | *** join/#asterisk ZeXr0 (~ZeXr0@ip216-239-95-218.vif.net) |
| 19:14.57 | ZeXr0 | I'd like to know, what would be the best way to play a message "To Continue in english, press 2", and then continu with the callflow if nothing is pressed |
| 19:18.40 | *** join/#asterisk Wildy (~simba@91.205.147.94) |
| 19:18.50 | *** join/#asterisk afink (~afink@204.26.87.226) |
| 19:19.11 | *** join/#asterisk Wildy (~simba@mas4-gw.pleer.ru) |
| 19:19.47 | afink | Hey guys I'm having trouble with a PRI t1 bouncing, seems to be losing the d-channel and Qwest wants to play the blame game. Can anyone help me get evidence that the problem is not asterisk. Or if it is? |
| 19:21.54 | *** join/#asterisk unspin (~unspin@S0106001451226d9c.vc.shawcable.net) |
| 19:22.13 | *** join/#asterisk {Repelex} (~{Repelex}@201-71-129-252-arpa.vsp.com.br) |
| 19:22.27 | wr| | WIMPy, using 1.6.2.10 now, gives me the same errors. |
| 19:25.15 | WIMPy | So what is it about? |
| 19:26.13 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 19:26.13 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 19:26.54 | wr| | chan_lcr.c:2381: error: dereferencing pointer to incomplete type. that line is: ast_playtones_start(ast, 0, ts->data, 1); |
| 19:27.19 | wr| | there are other "incompatible pointer type" warnings at lines calling ast_* functions as well |
| 19:29.44 | WIMPy | Your Asterisk is in working condition? |
| 19:30.10 | wr| | actually I don't know, just freshly compiled 2.6.2.10 |
| 19:30.17 | WIMPy | Can it find the headers? |
| 19:30.29 | WIMPy | And installed? |
| 19:30.31 | wr| | before that cleared all old headers |
| 19:30.32 | wr| | yes |
| 19:30.56 | wr| | starts up correctly |
| 19:31.05 | WIMPy | Called configure again? |
| 19:31.20 | wr| | yes |
| 19:31.30 | *** join/#asterisk [netman] (~netman@83.54.228.88) |
| 19:31.58 | *** join/#asterisk afink (~afink@204.26.87.226) |
| 19:32.10 | wr| | what's confusing is that ./configure seems to ignore my --prefix |
| 19:32.37 | wr| | but I doubt that is the problem |
| 19:34.07 | WIMPy | Are you using the current LCR? |
| 19:34.28 | afink | anyone have any insight on these errors: [Jul 26 14:35:11] ERROR[4291]: chan_dahdi.c:8744 dahdi_pri_error: Short write: 0/15 (Unknown error 500) and [Jul 26 14:35:11] ERROR[4291]: chan_dahdi.c:8744 dahdi_pri_error: Write to 41 failed: Unknown error 500 |
| 19:34.32 | WIMPy | I'm not sure how current the tarballs are. |
| 19:34.39 | wr| | 1.7 (from lcr-20100601.tar.gz) |
| 19:35.01 | wr| | hm. /me looking for a repo |
| 19:35.44 | *** join/#asterisk ManxPower (~manxpower@216.186.151.147) |
| 19:35.48 | wr| | hm. actually the tarball contains a git repo *g* unfortunately I'm not jolly@www.misdn.org... |
| 19:35.54 | ManxPower | ~answers |
| 19:35.55 | infobot | [~answers] Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki: http://www.voip-info.org & Docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: /path/to/src/asterisk/UPGRADE*.txt |
| 19:38.04 | wr| | WIMPy, do you know a public clone URL? I cant find any... |
| 19:38.13 | wr| | ah. found. |
| 19:38.15 | wr| | sorry |
| 19:38.44 | unspin | where can i find some decent quality sound files, english female (not Allison). French English or Spanish |
| 19:38.56 | unspin | been googling for a bit, haven't found too much |
| 19:40.56 | ZeXr0 | What would be the best way to play a message "To Continue in english, press 2", and then continue with the callflow if nothing is pressed |
| 19:41.57 | wr| | ZeXr0, maybe the timeout function? http://www.asteriskguru.com/tutorials/timeoutdigit_function.html but actually I don't know... |
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| 19:43.10 | wr| | WIMPy, latest lcr from git master compiling just fine. thank you. |
| 19:43.24 | *** join/#asterisk thansen (~thansen@c-98-202-28-239.hsd1.ut.comcast.net) |
| 19:43.29 | WIMPy | good |
| 19:43.53 | ZeXr0 | I guess Read(SendToEnglish, prompt) would do the work |
| 19:44.28 | goofy03 | hi i try to install asterisk with freepbx but when i make a call after numbering i get a weird ring not a real call |
| 19:44.38 | Qwell | goofy03: #freepbx |
| 19:45.03 | goofy03 | sorry ok |
| 19:46.47 | goofy03 | Channel 'DAHDI/2-1' sent into invalid extension 's' in context 'default', but no invalid handler do you know what this mean ? |
| 19:47.14 | Qwell | goofy03: It means you did something wrong. #freepbx |
| 19:48.44 | goofy03 | what is 's' extension ? i need a 63PO Droid |
| 19:49.21 | *** join/#asterisk Mhaddog (~Mhaddog@adsl-32-170-204.mia.bellsouth.net) |
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| 19:52.20 | *** join/#asterisk DrkShadow (~andrew@host-72-175-240-62.static.bresnan.net) |
| 19:52.35 | DrkShadow | how do I undo a shutdown request? ast_channel_alloc: Channel allocation failed: Refusing due to active shutdown |
| 19:52.48 | DrkShadow | I did restart gracefully.. I was under the impression it would NOT refuse incoming calls |
| 19:52.50 | *** join/#asterisk DelphiWorld (~Delphi@41.200.3.103) |
| 19:52.54 | DelphiWorld | hi |
| 19:52.57 | DelphiWorld | [TK]D-Fender: :P |
| 19:53.16 | Qwell | DrkShadow: that's the difference between "when convenient" and "gracefully" |
| 19:53.29 | Qwell | I actually don't think you can stop a shutdown |
| 19:53.39 | DrkShadow | *sigh* |
| 19:53.46 | Qwell | leifmadsen: truth? |
| 19:53.55 | *** join/#asterisk oc80z (oc80z@blea.ch) |
| 19:54.00 | leifmadsen | Qwell: I think there is actually a command to kill that... checking |
| 19:54.08 | DelphiWorld | hi leifmadsen |
| 19:54.14 | Qwell | I know there is in Linux, but... |
| 19:54.32 | leifmadsen | DrkShadow: core abort shutdown |
| 19:54.36 | Qwell | oh, huh. |
| 19:54.41 | leifmadsen | \o/ |
| 19:54.46 | leifmadsen | if it works... I have no idea :) |
| 19:54.53 | DrkShadow | damn. I JUST did a restart now cause we couldn't dial out ;-) thanks though. |
| 19:55.09 | Qwell | DrkShadow: ahh well. blame solar flares. |
| 19:55.16 | DrkShadow | hahaha, yep |
| 19:58.18 | *** join/#asterisk jeffgus (~jeffgus@2002:ad33:b504::1) |
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| 20:01.16 | *** part/#asterisk DelphiWorld (~Delphi@41.200.3.103) |
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| 20:19.08 | *** join/#asterisk Gershwin (~fake@unaffiliated/gershwin) |
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| 20:28.07 | *** join/#asterisk wcselby (60ee595a@gateway/web/freenode/ip.96.238.89.90) |
| 20:28.21 | wcselby | o/ |
| 20:29.51 | *** join/#asterisk brycebaril (~bbaril@sea02-v612-nat.marchex.com) |
| 20:34.09 | ZeXr0 | hum Is there a reason why AGI have a function SAY DATE but their is no equivalent function in Asterisk programming ? |
| 20:34.58 | WIMPy | sayunixtime |
| 20:36.06 | ZeXr0 | is date a real date, like 2010-05-05 or is it really a number of seconds ? |
| 20:38.51 | *** join/#asterisk [netman] (~netman@83.54.228.88) |
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| 20:46.48 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
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| 20:50.41 | *** join/#asterisk DelphiWorld (~Delphi@41.200.3.103) |
| 20:50.45 | *** part/#asterisk DelphiWorld (~Delphi@41.200.3.103) |
| 20:51.57 | carrar | *BURP* |
| 21:05.38 | *** join/#asterisk GabrielPiassetta (~gabriel@200.175.61.250) |
| 21:06.07 | GabrielPiassetta | hello, can i pass two or more files to "get option" in agi? |
| 21:07.36 | Qwell | GabrielPiassetta: I don't know what "get option" does, but when playing sound files, you can usually do file1&file2&file3 |
| 21:10.06 | *** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net) |
| 21:11.48 | *** join/#asterisk [netman] (~netman@83.52.208.225) |
| 21:15.31 | GabrielPiassetta | Qwell, in playback this works, but in get options no, get options behaves similar to stream file, its wait for user to make a choice in a ura |
| 21:26.41 | hardwire | any way to force no media for ringing? |
| 21:26.51 | hardwire | I want my servers completely out of the way for media |
| 21:27.14 | hardwire | call comes in via sip.. leaves via sip.. media for ringing still comes from asterisk.. no dial parameters being used and no answer on the asterisk box |
| 21:27.56 | *** join/#asterisk Godfather_ (~godfather@21.Red-88-7-2.staticIP.rima-tde.net) |
| 21:29.23 | hardwire | I have progressinband=never set |
| 21:29.55 | *** join/#asterisk unspin (~unspin@S010600031d02196a.vc.shawcable.net) |
| 21:31.14 | *** join/#asterisk DelphiWorld (~Delphi@41.200.3.103) |
| 21:31.15 | DelphiWorld | hi |
| 21:31.21 | DelphiWorld | how do i see my iax2 channel status? |
| 21:32.14 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:32.46 | DelphiWorld | how do i see my iax2 channel status? |
| 21:33.12 | russellb | *CLI> iax2 show channels |
| 21:34.32 | DelphiWorld | ok russellb |
| 21:34.36 | DelphiWorld | russellb: but this show channels |
| 21:34.45 | DelphiWorld | how do i see if my iax2 account is regged? |
| 21:35.09 | DelphiWorld | russellb: nm |
| 21:35.09 | russellb | *CLI> iax2 show registry |
| 21:35.09 | DelphiWorld | russellb: got it |
| 21:35.09 | DelphiWorld | russellb: iax2 show peers |
| 21:35.51 | DelphiWorld | thank you russellb |
| 21:35.53 | russellb | np |
| 21:36.43 | hardwire | this is so strange.. maybe I've just never noticed that even if I disable inband progress.. it's still there. |
| 21:38.35 | DelphiWorld | russellb: to reload config i use reload right? |
| 21:38.48 | *** join/#asterisk aidinb (~Aidin@wsip-98-190-45-225.oc.oc.cox.net) |
| 21:39.44 | russellb | yup |
| 21:39.50 | ManxPower | DelphiWorld, show peers shows CLIENTS registered TO your server. show registry shows YOU registered to OTHER servers. |
| 21:40.15 | DelphiWorld | thank you ManxPower and russellb |
| 21:40.40 | ManxPower | russell's answer is correct for your question. |
| 21:41.03 | DelphiWorld | ManxPower: yes...i was litle confused |
| 21:42.04 | joobie | burp |
| 21:44.45 | *** join/#asterisk Goshen (~Goshen@c-174-52-7-122.hsd1.ut.comcast.net) |
| 21:47.57 | DelphiWorld | could someone help me configure my asterisk |
| 21:48.07 | carrar | heh |
| 21:48.34 | carrar | Whats broken? |
| 21:48.40 | *** part/#asterisk ManxPower (~manxpower@216.186.151.147) |
| 21:50.23 | DelphiWorld | carrar: couldn't understand the config;) |
| 21:50.38 | carrar | Did you read the book? |
| 21:50.40 | [TK]D-Fender | pummels DelphiWorld with a hard-cover copy of THE BOOK |
| 21:50.40 | carrar | ~book |
| 21:50.52 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 21:51.02 | DelphiWorld | [TK]D-Fender: give me a Compiled html one no in pdf |
| 21:51.14 | DelphiWorld | [TK]D-Fender: is not fully accessible/readable by blind users |
| 21:51.16 | carrar | See above PDF link |
| 21:51.26 | DelphiWorld | Carlos_PHX1_: see my msg |
| 21:51.47 | [TK]D-Fender | DelphiWorld: Go pay someone to convert it for you |
| 21:51.49 | carrar | You can readh HTML or PDF? |
| 21:52.00 | carrar | Cause both exist out there |
| 21:52.04 | [TK]D-Fender | carrar: visually impared <- |
| 21:52.05 | DelphiWorld | [TK]D-Fender: give me a credit card to pay |
| 21:52.10 | DelphiWorld | Carlos_PHX1_: i can read html |
| 21:52.31 | DelphiWorld | [TK]D-Fender: if i can pay i don't wait for you to say it |
| 21:52.31 | [TK]D-Fender | DelphiWorld: Well the HTML link is right there |
| 21:53.20 | carrar | http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/index.html |
| 21:53.50 | carrar | TOO many options! |
| 21:57.26 | DelphiWorld | carrar: reading |
| 21:57.34 | carrar | Awesome!!!!!!!! |
| 21:58.10 | *** join/#asterisk pabelanger-lap (~pabelange@CPE0013f7abc09a-CM0013f7abc096.cpe.net.cable.rogers.com) |
| 21:58.21 | carrar | I'm gonna go to Costco then! |
| 21:59.30 | DelphiWorld | carrar: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_to_IAX.html |
| 22:03.44 | *** join/#asterisk verywiseman (~khaled@unaffiliated/verywiseman) |
| 22:09.05 | *** join/#asterisk aBs0lut30 (aBs0lut30@12.166.74.40) |
| 22:09.06 | *** part/#asterisk DelphiWorld (~Delphi@41.200.3.103) |
| 22:09.59 | aBs0lut30 | got a strange problem guys, on an outbound call over a sip trunk to a cisco gateway(non callmanager) the call goes out, but I never hear ringing, just dial, and after a few seconds your connected, any ideas? |
| 22:16.01 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 22:16.33 | anonymouz666 | very bizarre situation, a te420p installed on the system, and dahdi_tool shows OK for both spans without ANY cable connected. |
| 22:16.49 | anonymouz666 | anyone already seen that |
| 22:16.49 | anonymouz666 | ? |
| 22:18.15 | anonymouz666 | it should output RED... |
| 22:18.24 | anonymouz666 | I think this card is broken |
| 22:18.31 | leifmadsen | anonymouz666: time to call Digium |
| 22:20.05 | anonymouz666 | time to practice my english |
| 22:20.39 | *** join/#asterisk [netman] (~netman@83.52.208.225) |
| 22:20.42 | nightwalk | Ok, so I disabled callprogress, yet I *still* have users reporting being disconnected in the middle of their calls. Any ideas on the next most probable culprit? |
| 22:20.58 | anonymouz666 | Use callprogress=no |
| 22:21.05 | nightwalk | Already did that |
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| 22:26.07 | nightwalk | aBs0lut30: Maybe a side effect of answer()'ing in the dialplan? |
| 22:26.48 | *** join/#asterisk Mark22 (~mark@unaffiliated/mark21) |
| 22:26.49 | *** join/#asterisk romb (~romb@unaffiliated/romb-work/x-7222485) |
| 22:27.48 | Mark22 | after looking at multiple options I can't find a good solution for my problem :S I have a part from extensions.conf and from the log available at http://yourpaste.net/5848/ |
| 22:29.07 | Mark22 | the problem: the mobile phone (number starting with 06) goes to voicemail and with that voicemail the phone is answerd, however I want it to wait for me pressing a digit before it gets answerd for the caller (in this case sip account 1010) |
| 22:29.36 | Mark22 | I want something that I could use with multiple numbers (multiple mobile phones/sip accounts at the same time) |
| 22:29.52 | Mark22 | is there a solution for it or am I looking in the wrong direction? |
| 22:34.09 | *** join/#asterisk prgmrchris (~chris@66.9.61.162) |
| 22:41.46 | nightwalk | Mark22: That wasn't really very clear to me |
| 22:43.17 | Mark22 | I was testing if I could call a number (in this example 9000) so a mobile phone rings and when I pickup I need to press a key before the caller knows that I did pick up that mobile phone |
| 22:43.46 | Mark22 | when the mobile phone is already in use it goes directly to the voicemail on that mobile phone and that is something I don't want |
| 22:44.16 | Mark22 | so the voicemail on the mobile phone is hopefully never used (or at least the caller doesn't know it is used) |
| 22:47.41 | DogBoy | what do you want to happen instead |
| 22:47.55 | *** join/#asterisk russellb_ (~russell@asterisk/digium-open-source-team-lead/russellb) |
| 22:47.56 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 22:49.05 | Mark22 | in the future I'll add at least one sip account and probably 3 sip accounts and an extra mobile phone so they should ring and after 30 seconds (or another time) it should retry to see if the phone is available or not |
| 22:49.34 | Mark22 | even going to another extension in asterisk is acceptable |
| 22:50.08 | Mark22 | as long as the voicemail from the mobile phone isn't given to the original caller (and I can't disable that f*cking voicemail) |
| 22:56.48 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 23:06.04 | [TK]D-Fender | Mark22: Problem is you are performing an SASO in trying to create an IVR in your macro. This is NOT how you do it. Use READ() |
| 23:06.06 | [TK]D-Fender | ~saso |
| 23:06.22 | [TK]D-Fender | ~SOSO |
| 23:06.23 | infobot | [~soso] Shoot-On-Sight Offense |
| 23:06.29 | [TK]D-Fender | There we go |
| 23:06.55 | [TK]D-Fender | Mark22: You are "running out of s" on your macro and it jsut falls through. it won't wait. Wrong approach. Use Read() |
| 23:08.15 | Mark22 | looking at it now |
| 23:12.50 | *** join/#asterisk evilcoder (~shani@67.230.191.198) |
| 23:17.12 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.93) |
| 23:19.57 | *** join/#asterisk Deeewayne (~dwayne@c-71-207-214-190.hsd1.al.comcast.net) |
| 23:19.57 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 23:22.16 | nightwalk | Any other probable culprits for dropped calls, aside from callprogress? |
| 23:24.34 | nightwalk | Only thing I'm seeing are the MFC/R2 options, and if I read correctly, that doesn't apply since I'm in the US |
| 23:28.51 | [TK]D-Fender | nightwalk: And you haven't told us what you ARE using. |
| 23:29.12 | nightwalk | tdm400 compatible card, 1 FXS, 3 FXO. |
| 23:30.07 | nightwalk | FXO-FXO calls are fine, but FXO-FXS calls drop every once in a while. |
| 23:30.07 | nightwalk | Maybe...busydetect? I have busycount set to 8 already. Any bad side effects from just disabling it? |
| 23:32.25 | Mark22 | [TK]D-Fender: thank you for pointing in the right direction now it is working (if someone wants a copy from what I've now just ask) |
| 23:32.40 | [TK]D-Fender | callprogress=yes is the usual culprit. The other is odd spikes triggering a CPC. |
| 23:32.49 | [TK]D-Fender | ~CDS |
| 23:32.50 | infobot | [~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line. |
| 23:32.51 | [TK]D-Fender | rather |
| 23:34.29 | nightwalk | callprogress is already disabled, and if I'm not mistaken (which I very well may be), the cds options are in with the MFC/R2 options, which don't apply since this is in the US |
| 23:34.53 | nightwalk | Err..*apparently* don't apply, rather |
| 23:35.29 | [TK]D-Fender | nightwalk: CDS is a standard analog think |
| 23:35.31 | [TK]D-Fender | thing |
| 23:35.40 | [TK]D-Fender | nightwalk: Please never ever mention R2 again, OK? |
| 23:35.53 | [TK]D-Fender | nightwalk: Wrong tree. Nothing applicable. |
| 23:36.14 | nightwalk | No problem. I'm American so who *cares*, about the rest of the world? :) |
| 23:37.37 | *** join/#asterisk root52 (~root52@ip24-252-251-246.cl.ri.cox.net) |
| 23:37.43 | nightwalk | So, if I have this straight, you're saying the telco may just suck? |
| 23:37.46 | [TK]D-Fender | nightwalk: Thats the spirit... |
| 23:38.22 | [TK]D-Fender | nightwalk: Could be.... |
| 23:38.28 | nightwalk | Because that's a distinct possibility out here in BFA.... |
| 23:39.52 | [TK]D-Fender | times up... out for a bit, BBL |
| 23:41.34 | root52 | Good Day all. I have a problem with an asterisk server randomly crashing. So in the safe_asterisk man page it says that it will cause asterisk to "dump core" if it were to crash. along with trying to restart it. I have got core dumps in the past by re complying asterisk with (i think it was...) the CORE_DUMP option checked. So am I to assume that if I start asterisk with safe_asterisk and it crashes I will end up with a dump? |
| 23:41.34 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 23:43.29 | nightwalk | [TK]D-Fender: Well, I upped busycount from 8 to 12, and set busypattern (with the default of 500, 500). That's the only things I see here config-wise that seem like they kinda, sorta could cause this problem. |
| 23:44.21 | nightwalk | root52: That's probably dependent on your system allowing core dumps in addition to the asterisk-specific options |
| 23:56.50 | Mark22 | root52: what asterisk version are you using? please provide more information |